Realtime
realtime.connect(RealtimeConnectParams**kwargs)
**** ``
The Realtime API enables you to build low-latency, multi-modal conversational experiences. It currently supports text and audio as both input and output, as well as function calling.
Some notable benefits of the API include:
- Native speech-to-speech: Skipping an intermediate text format means low latency and nuanced output.
- Natural, steerable voices: The models have natural inflection and can laugh, whisper, and adhere to tone direction.
- Simultaneous multimodal output: Text is useful for moderation; faster-than-realtime audio ensures stable playback.
The Realtime API is a stateful, event-based API that communicates over a WebSocket.
Parameters
-
call_id: Optional[str] -
model: Optional[str]
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client.realtime.connect()
Domain Types
Audio Transcription
-
class AudioTranscription: …-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
Conversation Created Event
-
class ConversationCreatedEvent: …Returned when a conversation is created. Emitted right after session creation.
-
conversation: ConversationThe conversation resource.
-
id: Optional[str]The unique ID of the conversation.
-
object: Optional[Literal["realtime.conversation"]]The object type, must be
realtime.conversation."realtime.conversation"
-
-
event_id: strThe unique ID of the server event.
-
type: Literal["conversation.created"]The event type, must be
conversation.created."conversation.created"
-
Conversation Item
-
ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
Conversation Item Added
-
class ConversationItemAdded: …Sent by the server when an Item is added to the default Conversation. This can happen in several cases:
- When the client sends a
conversation.item.createevent. - When the input audio buffer is committed. In this case the item will be a user message containing the audio from the buffer.
- When the model is generating a Response. In this case the
conversation.item.addedevent will be sent when the model starts generating a specific Item, and thus it will not yet have any content (andstatuswill bein_progress).
The event will include the full content of the Item (except when model is generating a Response) except for audio data, which can be retrieved separately with a
conversation.item.retrieveevent if necessary.-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.added"]The event type, must be
conversation.item.added."conversation.item.added"
-
previous_item_id: Optional[str]The ID of the item that precedes this one, if any. This is used to maintain ordering when items are inserted.
- When the client sends a
Conversation Item Create Event
-
class ConversationItemCreateEvent: …Add a new Item to the Conversation's context, including messages, function calls, and function call responses. This event can be used both to populate a "history" of the conversation and to add new items mid-stream, but has the current limitation that it cannot populate assistant audio messages.
If successful, the server will respond with a
conversation.item.createdevent, otherwise anerrorevent will be sent.-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.create"]The event type, must be
conversation.item.create."conversation.item.create"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
previous_item_id: Optional[str]The ID of the preceding item after which the new item will be inserted. If not set, the new item will be appended to the end of the conversation.
If set to
root, the new item will be added to the beginning of the conversation.If set to an existing ID, it allows an item to be inserted mid-conversation. If the ID cannot be found, an error will be returned and the item will not be added.
-
Conversation Item Created Event
-
class ConversationItemCreatedEvent: …Returned when a conversation item is created. There are several scenarios that produce this event:
-
The server is generating a Response, which if successful will produce either one or two Items, which will be of type
message(roleassistant) or typefunction_call. -
The input audio buffer has been committed, either by the client or the server (in
server_vadmode). The server will take the content of the input audio buffer and add it to a new user message Item. -
The client has sent a
conversation.item.createevent to add a new Item to the Conversation. -
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.created"]The event type, must be
conversation.item.created."conversation.item.created"
-
previous_item_id: Optional[str]The ID of the preceding item in the Conversation context, allows the client to understand the order of the conversation. Can be
nullif the item has no predecessor.
-
Conversation Item Delete Event
-
class ConversationItemDeleteEvent: …Send this event when you want to remove any item from the conversation history. The server will respond with a
conversation.item.deletedevent, unless the item does not exist in the conversation history, in which case the server will respond with an error.-
item_id: strThe ID of the item to delete.
-
type: Literal["conversation.item.delete"]The event type, must be
conversation.item.delete."conversation.item.delete"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Conversation Item Deleted Event
-
class ConversationItemDeletedEvent: …Returned when an item in the conversation is deleted by the client with a
conversation.item.deleteevent. This event is used to synchronize the server's understanding of the conversation history with the client's view.-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item that was deleted.
-
type: Literal["conversation.item.deleted"]The event type, must be
conversation.item.deleted."conversation.item.deleted"
-
Conversation Item Done
-
class ConversationItemDone: …Returned when a conversation item is finalized.
The event will include the full content of the Item except for audio data, which can be retrieved separately with a
conversation.item.retrieveevent if needed.-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.done"]The event type, must be
conversation.item.done."conversation.item.done"
-
previous_item_id: Optional[str]The ID of the item that precedes this one, if any. This is used to maintain ordering when items are inserted.
-
Conversation Item Input Audio Transcription Completed Event
-
class ConversationItemInputAudioTranscriptionCompletedEvent: …This event is the output of audio transcription for user audio written to the user audio buffer. Transcription begins when the input audio buffer is committed by the client or server (when VAD is enabled). Transcription runs asynchronously with Response creation, so this event may come before or after the Response events.
Realtime API models accept audio natively, and thus input transcription is a separate process run on a separate ASR (Automatic Speech Recognition) model. The transcript may diverge somewhat from the model's interpretation, and should be treated as a rough guide.
-
content_index: intThe index of the content part containing the audio.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the audio that is being transcribed.
-
transcript: strThe transcribed text.
-
type: Literal["conversation.item.input_audio_transcription.completed"]The event type, must be
conversation.item.input_audio_transcription.completed."conversation.item.input_audio_transcription.completed"
-
usage: UsageUsage statistics for the transcription, this is billed according to the ASR model's pricing rather than the realtime model's pricing.
-
class UsageTranscriptTextUsageTokens: …Usage statistics for models billed by token usage.
-
input_tokens: intNumber of input tokens billed for this request.
-
output_tokens: intNumber of output tokens generated.
-
total_tokens: intTotal number of tokens used (input + output).
-
type: Literal["tokens"]The type of the usage object. Always
tokensfor this variant."tokens"
-
input_token_details: Optional[UsageTranscriptTextUsageTokensInputTokenDetails]Details about the input tokens billed for this request.
-
audio_tokens: Optional[int]Number of audio tokens billed for this request.
-
text_tokens: Optional[int]Number of text tokens billed for this request.
-
-
-
class UsageTranscriptTextUsageDuration: …Usage statistics for models billed by audio input duration.
-
seconds: floatDuration of the input audio in seconds.
-
type: Literal["duration"]The type of the usage object. Always
durationfor this variant."duration"
-
-
-
logprobs: Optional[List[LogProbProperties]]The log probabilities of the transcription.
-
token: strThe token that was used to generate the log probability.
-
bytes: List[int]The bytes that were used to generate the log probability.
-
logprob: floatThe log probability of the token.
-
-
Conversation Item Input Audio Transcription Delta Event
-
class ConversationItemInputAudioTranscriptionDeltaEvent: …Returned when the text value of an input audio transcription content part is updated with incremental transcription results.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the audio that is being transcribed.
-
type: Literal["conversation.item.input_audio_transcription.delta"]The event type, must be
conversation.item.input_audio_transcription.delta."conversation.item.input_audio_transcription.delta"
-
content_index: Optional[int]The index of the content part in the item's content array.
-
delta: Optional[str]The text delta.
-
logprobs: Optional[List[LogProbProperties]]The log probabilities of the transcription. These can be enabled by configurating the session with
"include": ["item.input_audio_transcription.logprobs"]. Each entry in the array corresponds a log probability of which token would be selected for this chunk of transcription. This can help to identify if it was possible there were multiple valid options for a given chunk of transcription.-
token: strThe token that was used to generate the log probability.
-
bytes: List[int]The bytes that were used to generate the log probability.
-
logprob: floatThe log probability of the token.
-
-
Conversation Item Input Audio Transcription Failed Event
-
class ConversationItemInputAudioTranscriptionFailedEvent: …Returned when input audio transcription is configured, and a transcription request for a user message failed. These events are separate from other
errorevents so that the client can identify the related Item.-
content_index: intThe index of the content part containing the audio.
-
error: ErrorDetails of the transcription error.
-
code: Optional[str]Error code, if any.
-
message: Optional[str]A human-readable error message.
-
param: Optional[str]Parameter related to the error, if any.
-
type: Optional[str]The type of error.
-
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item.
-
type: Literal["conversation.item.input_audio_transcription.failed"]The event type, must be
conversation.item.input_audio_transcription.failed."conversation.item.input_audio_transcription.failed"
-
Conversation Item Input Audio Transcription Segment
-
class ConversationItemInputAudioTranscriptionSegment: …Returned when an input audio transcription segment is identified for an item.
-
id: strThe segment identifier.
-
content_index: intThe index of the input audio content part within the item.
-
end: floatEnd time of the segment in seconds.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the input audio content.
-
speaker: strThe detected speaker label for this segment.
-
start: floatStart time of the segment in seconds.
-
text: strThe text for this segment.
-
type: Literal["conversation.item.input_audio_transcription.segment"]The event type, must be
conversation.item.input_audio_transcription.segment."conversation.item.input_audio_transcription.segment"
-
Conversation Item Retrieve Event
-
class ConversationItemRetrieveEvent: …Send this event when you want to retrieve the server's representation of a specific item in the conversation history. This is useful, for example, to inspect user audio after noise cancellation and VAD. The server will respond with a
conversation.item.retrievedevent, unless the item does not exist in the conversation history, in which case the server will respond with an error.-
item_id: strThe ID of the item to retrieve.
-
type: Literal["conversation.item.retrieve"]The event type, must be
conversation.item.retrieve."conversation.item.retrieve"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Conversation Item Truncate Event
-
class ConversationItemTruncateEvent: …Send this event to truncate a previous assistant message’s audio. The server will produce audio faster than realtime, so this event is useful when the user interrupts to truncate audio that has already been sent to the client but not yet played. This will synchronize the server's understanding of the audio with the client's playback.
Truncating audio will delete the server-side text transcript to ensure there is not text in the context that hasn't been heard by the user.
If successful, the server will respond with a
conversation.item.truncatedevent.-
audio_end_ms: intInclusive duration up to which audio is truncated, in milliseconds. If the audio_end_ms is greater than the actual audio duration, the server will respond with an error.
-
content_index: intThe index of the content part to truncate. Set this to
0. -
item_id: strThe ID of the assistant message item to truncate. Only assistant message items can be truncated.
-
type: Literal["conversation.item.truncate"]The event type, must be
conversation.item.truncate."conversation.item.truncate"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Conversation Item Truncated Event
-
class ConversationItemTruncatedEvent: …Returned when an earlier assistant audio message item is truncated by the client with a
conversation.item.truncateevent. This event is used to synchronize the server's understanding of the audio with the client's playback.This action will truncate the audio and remove the server-side text transcript to ensure there is no text in the context that hasn't been heard by the user.
-
audio_end_ms: intThe duration up to which the audio was truncated, in milliseconds.
-
content_index: intThe index of the content part that was truncated.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the assistant message item that was truncated.
-
type: Literal["conversation.item.truncated"]The event type, must be
conversation.item.truncated."conversation.item.truncated"
-
Conversation Item With Reference
-
class ConversationItemWithReference: …The item to add to the conversation.
-
id: Optional[str]For an item of type (
message|function_call|function_call_output) this field allows the client to assign the unique ID of the item. It is not required because the server will generate one if not provided.For an item of type
item_reference, this field is required and is a reference to any item that has previously existed in the conversation. -
arguments: Optional[str]The arguments of the function call (for
function_callitems). -
call_id: Optional[str]The ID of the function call (for
function_callandfunction_call_outputitems). If passed on afunction_call_outputitem, the server will check that afunction_callitem with the same ID exists in the conversation history. -
content: Optional[List[Content]]The content of the message, applicable for
messageitems.-
Message items of role
systemsupport onlyinput_textcontent -
Message items of role
usersupportinput_textandinput_audiocontent -
Message items of role
assistantsupporttextcontent. -
id: Optional[str]ID of a previous conversation item to reference (for
item_referencecontent types inresponse.createevents). These can reference both client and server created items. -
audio: Optional[str]Base64-encoded audio bytes, used for
input_audiocontent type. -
text: Optional[str]The text content, used for
input_textandtextcontent types. -
transcript: Optional[str]The transcript of the audio, used for
input_audiocontent type. -
type: Optional[Literal["input_text", "input_audio", "item_reference", "text"]]The content type (
input_text,input_audio,item_reference,text).-
"input_text" -
"input_audio" -
"item_reference" -
"text"
-
-
-
name: Optional[str]The name of the function being called (for
function_callitems). -
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item."realtime.item"
-
output: Optional[str]The output of the function call (for
function_call_outputitems). -
role: Optional[Literal["user", "assistant", "system"]]The role of the message sender (
user,assistant,system), only applicable formessageitems.-
"user" -
"assistant" -
"system"
-
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item (
completed,incomplete,in_progress). These have no effect on the conversation, but are accepted for consistency with theconversation.item.createdevent.-
"completed" -
"incomplete" -
"in_progress"
-
-
type: Optional[Literal["message", "function_call", "function_call_output", "item_reference"]]The type of the item (
message,function_call,function_call_output,item_reference).-
"message" -
"function_call" -
"function_call_output" -
"item_reference"
-
-
Input Audio Buffer Append Event
-
class InputAudioBufferAppendEvent: …Send this event to append audio bytes to the input audio buffer. The audio buffer is temporary storage you can write to and later commit. A "commit" will create a new user message item in the conversation history from the buffer content and clear the buffer. Input audio transcription (if enabled) will be generated when the buffer is committed.
If VAD is enabled the audio buffer is used to detect speech and the server will decide when to commit. When Server VAD is disabled, you must commit the audio buffer manually. Input audio noise reduction operates on writes to the audio buffer.
The client may choose how much audio to place in each event up to a maximum of 15 MiB, for example streaming smaller chunks from the client may allow the VAD to be more responsive. Unlike most other client events, the server will not send a confirmation response to this event.
-
audio: strBase64-encoded audio bytes. This must be in the format specified by the
input_audio_formatfield in the session configuration. -
type: Literal["input_audio_buffer.append"]The event type, must be
input_audio_buffer.append."input_audio_buffer.append"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Input Audio Buffer Clear Event
-
class InputAudioBufferClearEvent: …Send this event to clear the audio bytes in the buffer. The server will respond with an
input_audio_buffer.clearedevent.-
type: Literal["input_audio_buffer.clear"]The event type, must be
input_audio_buffer.clear."input_audio_buffer.clear"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Input Audio Buffer Cleared Event
-
class InputAudioBufferClearedEvent: …Returned when the input audio buffer is cleared by the client with a
input_audio_buffer.clearevent.-
event_id: strThe unique ID of the server event.
-
type: Literal["input_audio_buffer.cleared"]The event type, must be
input_audio_buffer.cleared."input_audio_buffer.cleared"
-
Input Audio Buffer Commit Event
-
class InputAudioBufferCommitEvent: …Send this event to commit the user input audio buffer, which will create a new user message item in the conversation. This event will produce an error if the input audio buffer is empty. When in Server VAD mode, the client does not need to send this event, the server will commit the audio buffer automatically.
Committing the input audio buffer will trigger input audio transcription (if enabled in session configuration), but it will not create a response from the model. The server will respond with an
input_audio_buffer.committedevent.-
type: Literal["input_audio_buffer.commit"]The event type, must be
input_audio_buffer.commit."input_audio_buffer.commit"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Input Audio Buffer Committed Event
-
class InputAudioBufferCommittedEvent: …Returned when an input audio buffer is committed, either by the client or automatically in server VAD mode. The
item_idproperty is the ID of the user message item that will be created, thus aconversation.item.createdevent will also be sent to the client.-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created.
-
type: Literal["input_audio_buffer.committed"]The event type, must be
input_audio_buffer.committed."input_audio_buffer.committed"
-
previous_item_id: Optional[str]The ID of the preceding item after which the new item will be inserted. Can be
nullif the item has no predecessor.
-
Input Audio Buffer Dtmf Event Received Event
-
class InputAudioBufferDtmfEventReceivedEvent: …SIP Only: Returned when an DTMF event is received. A DTMF event is a message that represents a telephone keypad press (0–9, *, #, A–D). The
eventproperty is the keypad that the user press. Thereceived_atis the UTC Unix Timestamp that the server received the event.-
event: strThe telephone keypad that was pressed by the user.
-
received_at: intUTC Unix Timestamp when DTMF Event was received by server.
-
type: Literal["input_audio_buffer.dtmf_event_received"]The event type, must be
input_audio_buffer.dtmf_event_received."input_audio_buffer.dtmf_event_received"
-
Input Audio Buffer Speech Started Event
-
class InputAudioBufferSpeechStartedEvent: …Sent by the server when in
server_vadmode to indicate that speech has been detected in the audio buffer. This can happen any time audio is added to the buffer (unless speech is already detected). The client may want to use this event to interrupt audio playback or provide visual feedback to the user.The client should expect to receive a
input_audio_buffer.speech_stoppedevent when speech stops. Theitem_idproperty is the ID of the user message item that will be created when speech stops and will also be included in theinput_audio_buffer.speech_stoppedevent (unless the client manually commits the audio buffer during VAD activation).-
audio_start_ms: intMilliseconds from the start of all audio written to the buffer during the session when speech was first detected. This will correspond to the beginning of audio sent to the model, and thus includes the
prefix_padding_msconfigured in the Session. -
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created when speech stops.
-
type: Literal["input_audio_buffer.speech_started"]The event type, must be
input_audio_buffer.speech_started."input_audio_buffer.speech_started"
-
Input Audio Buffer Speech Stopped Event
-
class InputAudioBufferSpeechStoppedEvent: …Returned in
server_vadmode when the server detects the end of speech in the audio buffer. The server will also send anconversation.item.createdevent with the user message item that is created from the audio buffer.-
audio_end_ms: intMilliseconds since the session started when speech stopped. This will correspond to the end of audio sent to the model, and thus includes the
min_silence_duration_msconfigured in the Session. -
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created.
-
type: Literal["input_audio_buffer.speech_stopped"]The event type, must be
input_audio_buffer.speech_stopped."input_audio_buffer.speech_stopped"
-
Input Audio Buffer Timeout Triggered
-
class InputAudioBufferTimeoutTriggered: …Returned when the Server VAD timeout is triggered for the input audio buffer. This is configured with
idle_timeout_msin theturn_detectionsettings of the session, and it indicates that there hasn't been any speech detected for the configured duration.The
audio_start_msandaudio_end_msfields indicate the segment of audio after the last model response up to the triggering time, as an offset from the beginning of audio written to the input audio buffer. This means it demarcates the segment of audio that was silent and the difference between the start and end values will roughly match the configured timeout.The empty audio will be committed to the conversation as an
input_audioitem (there will be ainput_audio_buffer.committedevent) and a model response will be generated. There may be speech that didn't trigger VAD but is still detected by the model, so the model may respond with something relevant to the conversation or a prompt to continue speaking.-
audio_end_ms: intMillisecond offset of audio written to the input audio buffer at the time the timeout was triggered.
-
audio_start_ms: intMillisecond offset of audio written to the input audio buffer that was after the playback time of the last model response.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item associated with this segment.
-
type: Literal["input_audio_buffer.timeout_triggered"]The event type, must be
input_audio_buffer.timeout_triggered."input_audio_buffer.timeout_triggered"
-
Log Prob Properties
-
class LogProbProperties: …A log probability object.
-
token: strThe token that was used to generate the log probability.
-
bytes: List[int]The bytes that were used to generate the log probability.
-
logprob: floatThe log probability of the token.
-
Mcp List Tools Completed
-
class McpListToolsCompleted: …Returned when listing MCP tools has completed for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.completed"]The event type, must be
mcp_list_tools.completed."mcp_list_tools.completed"
-
Mcp List Tools Failed
-
class McpListToolsFailed: …Returned when listing MCP tools has failed for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.failed"]The event type, must be
mcp_list_tools.failed."mcp_list_tools.failed"
-
Mcp List Tools In Progress
-
class McpListToolsInProgress: …Returned when listing MCP tools is in progress for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.in_progress"]The event type, must be
mcp_list_tools.in_progress."mcp_list_tools.in_progress"
-
Noise Reduction Type
-
Literal["near_field", "far_field"]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
Output Audio Buffer Clear Event
-
class OutputAudioBufferClearEvent: …WebRTC/SIP Only: Emit to cut off the current audio response. This will trigger the server to stop generating audio and emit a
output_audio_buffer.clearedevent. This event should be preceded by aresponse.cancelclient event to stop the generation of the current response. Learn more.-
type: Literal["output_audio_buffer.clear"]The event type, must be
output_audio_buffer.clear."output_audio_buffer.clear"
-
event_id: Optional[str]The unique ID of the client event used for error handling.
-
Rate Limits Updated Event
-
class RateLimitsUpdatedEvent: …Emitted at the beginning of a Response to indicate the updated rate limits. When a Response is created some tokens will be "reserved" for the output tokens, the rate limits shown here reflect that reservation, which is then adjusted accordingly once the Response is completed.
-
event_id: strThe unique ID of the server event.
-
rate_limits: List[RateLimit]List of rate limit information.
-
limit: Optional[int]The maximum allowed value for the rate limit.
-
name: Optional[Literal["requests", "tokens"]]The name of the rate limit (
requests,tokens).-
"requests" -
"tokens"
-
-
remaining: Optional[int]The remaining value before the limit is reached.
-
reset_seconds: Optional[float]Seconds until the rate limit resets.
-
-
type: Literal["rate_limits.updated"]The event type, must be
rate_limits.updated."rate_limits.updated"
-
Realtime Audio Config
-
class RealtimeAudioConfig: …Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
Realtime Audio Config Input
-
class RealtimeAudioConfigInput: …-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
Realtime Audio Config Output
-
class RealtimeAudioConfigOutput: …-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
Realtime Audio Formats
-
RealtimeAudioFormatsThe PCM audio format. Only a 24kHz sample rate is supported.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
Realtime Audio Input Turn Detection
-
Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
Realtime Client Event
-
RealtimeClientEventA realtime client event.
-
class ConversationItemCreateEvent: …Add a new Item to the Conversation's context, including messages, function calls, and function call responses. This event can be used both to populate a "history" of the conversation and to add new items mid-stream, but has the current limitation that it cannot populate assistant audio messages.
If successful, the server will respond with a
conversation.item.createdevent, otherwise anerrorevent will be sent.-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.create"]The event type, must be
conversation.item.create."conversation.item.create"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
previous_item_id: Optional[str]The ID of the preceding item after which the new item will be inserted. If not set, the new item will be appended to the end of the conversation.
If set to
root, the new item will be added to the beginning of the conversation.If set to an existing ID, it allows an item to be inserted mid-conversation. If the ID cannot be found, an error will be returned and the item will not be added.
-
-
class ConversationItemDeleteEvent: …Send this event when you want to remove any item from the conversation history. The server will respond with a
conversation.item.deletedevent, unless the item does not exist in the conversation history, in which case the server will respond with an error.-
item_id: strThe ID of the item to delete.
-
type: Literal["conversation.item.delete"]The event type, must be
conversation.item.delete."conversation.item.delete"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class ConversationItemRetrieveEvent: …Send this event when you want to retrieve the server's representation of a specific item in the conversation history. This is useful, for example, to inspect user audio after noise cancellation and VAD. The server will respond with a
conversation.item.retrievedevent, unless the item does not exist in the conversation history, in which case the server will respond with an error.-
item_id: strThe ID of the item to retrieve.
-
type: Literal["conversation.item.retrieve"]The event type, must be
conversation.item.retrieve."conversation.item.retrieve"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class ConversationItemTruncateEvent: …Send this event to truncate a previous assistant message’s audio. The server will produce audio faster than realtime, so this event is useful when the user interrupts to truncate audio that has already been sent to the client but not yet played. This will synchronize the server's understanding of the audio with the client's playback.
Truncating audio will delete the server-side text transcript to ensure there is not text in the context that hasn't been heard by the user.
If successful, the server will respond with a
conversation.item.truncatedevent.-
audio_end_ms: intInclusive duration up to which audio is truncated, in milliseconds. If the audio_end_ms is greater than the actual audio duration, the server will respond with an error.
-
content_index: intThe index of the content part to truncate. Set this to
0. -
item_id: strThe ID of the assistant message item to truncate. Only assistant message items can be truncated.
-
type: Literal["conversation.item.truncate"]The event type, must be
conversation.item.truncate."conversation.item.truncate"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class InputAudioBufferAppendEvent: …Send this event to append audio bytes to the input audio buffer. The audio buffer is temporary storage you can write to and later commit. A "commit" will create a new user message item in the conversation history from the buffer content and clear the buffer. Input audio transcription (if enabled) will be generated when the buffer is committed.
If VAD is enabled the audio buffer is used to detect speech and the server will decide when to commit. When Server VAD is disabled, you must commit the audio buffer manually. Input audio noise reduction operates on writes to the audio buffer.
The client may choose how much audio to place in each event up to a maximum of 15 MiB, for example streaming smaller chunks from the client may allow the VAD to be more responsive. Unlike most other client events, the server will not send a confirmation response to this event.
-
audio: strBase64-encoded audio bytes. This must be in the format specified by the
input_audio_formatfield in the session configuration. -
type: Literal["input_audio_buffer.append"]The event type, must be
input_audio_buffer.append."input_audio_buffer.append"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class InputAudioBufferClearEvent: …Send this event to clear the audio bytes in the buffer. The server will respond with an
input_audio_buffer.clearedevent.-
type: Literal["input_audio_buffer.clear"]The event type, must be
input_audio_buffer.clear."input_audio_buffer.clear"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class OutputAudioBufferClearEvent: …WebRTC/SIP Only: Emit to cut off the current audio response. This will trigger the server to stop generating audio and emit a
output_audio_buffer.clearedevent. This event should be preceded by aresponse.cancelclient event to stop the generation of the current response. Learn more.-
type: Literal["output_audio_buffer.clear"]The event type, must be
output_audio_buffer.clear."output_audio_buffer.clear"
-
event_id: Optional[str]The unique ID of the client event used for error handling.
-
-
class InputAudioBufferCommitEvent: …Send this event to commit the user input audio buffer, which will create a new user message item in the conversation. This event will produce an error if the input audio buffer is empty. When in Server VAD mode, the client does not need to send this event, the server will commit the audio buffer automatically.
Committing the input audio buffer will trigger input audio transcription (if enabled in session configuration), but it will not create a response from the model. The server will respond with an
input_audio_buffer.committedevent.-
type: Literal["input_audio_buffer.commit"]The event type, must be
input_audio_buffer.commit."input_audio_buffer.commit"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class ResponseCancelEvent: …Send this event to cancel an in-progress response. The server will respond with a
response.doneevent with a status ofresponse.status=cancelled. If there is no response to cancel, the server will respond with an error. It's safe to callresponse.canceleven if no response is in progress, an error will be returned the session will remain unaffected.-
type: Literal["response.cancel"]The event type, must be
response.cancel."response.cancel"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
response_id: Optional[str]A specific response ID to cancel - if not provided, will cancel an in-progress response in the default conversation.
-
-
class ResponseCreateEvent: …This event instructs the server to create a Response, which means triggering model inference. When in Server VAD mode, the server will create Responses automatically.
A Response will include at least one Item, and may have two, in which case the second will be a function call. These Items will be appended to the conversation history by default.
The server will respond with a
response.createdevent, events for Items and content created, and finally aresponse.doneevent to indicate the Response is complete.The
response.createevent includes inference configuration likeinstructionsandtools. If these are set, they will override the Session's configuration for this Response only.Responses can be created out-of-band of the default Conversation, meaning that they can have arbitrary input, and it's possible to disable writing the output to the Conversation. Only one Response can write to the default Conversation at a time, but otherwise multiple Responses can be created in parallel. The
metadatafield is a good way to disambiguate multiple simultaneous Responses.Clients can set
conversationtononeto create a Response that does not write to the default Conversation. Arbitrary input can be provided with theinputfield, which is an array accepting raw Items and references to existing Items.-
type: Literal["response.create"]The event type, must be
response.create."response.create"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
response: Optional[RealtimeResponseCreateParams]Create a new Realtime response with these parameters
-
audio: Optional[RealtimeResponseCreateAudioOutput]Configuration for audio input and output.
-
output: Optional[Output]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[OutputVoice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class OutputVoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
conversation: Optional[Union[str, Literal["auto", "none"], null]]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
str -
Literal["auto", "none"]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
"auto" -
"none"
-
-
-
input: Optional[List[ConversationItem]]Input items to include in the prompt for the model. Using this field creates a new context for this Response instead of using the default conversation. An empty array
[]will clear the context for this Response. Note that this can include references to items that previously appeared in the session using their id.-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior. Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class RealtimeResponseCreateMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[AllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class AllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[RequireApproval]Specify which of the MCP server's tools require approval.
-
class RequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[RequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[RequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
-
-
class SessionUpdateEvent: …Send this event to update the session’s configuration. The client may send this event at any time to update any field except for
voiceandmodel.voicecan be updated only if there have been no other audio outputs yet.When the server receives a
session.update, it will respond with asession.updatedevent showing the full, effective configuration. Only the fields that are present in thesession.updateare updated. To clear a field likeinstructions, pass an empty string. To clear a field liketools, pass an empty array. To clear a field liketurn_detection, passnull.-
session: SessionUpdate the Realtime session. Choose either a realtime session or a transcription session.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2. -
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: … -
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
-
type: Literal["session.update"]The event type, must be
session.update."session.update"
-
event_id: Optional[str]Optional client-generated ID used to identify this event. This is an arbitrary string that a client may assign. It will be passed back if there is an error with the event, but the corresponding
session.updatedevent will not include it.
-
-
Realtime Conversation Item Assistant Message
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
Realtime Conversation Item Function Call
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
Realtime Conversation Item Function Call Output
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
Realtime Conversation Item System Message
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
Realtime Conversation Item User Message
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
Realtime Error
-
class RealtimeError: …Details of the error.
-
message: strA human-readable error message.
-
type: strThe type of error (e.g., "invalid_request_error", "server_error").
-
code: Optional[str]Error code, if any.
-
event_id: Optional[str]The event_id of the client event that caused the error, if applicable.
-
param: Optional[str]Parameter related to the error, if any.
-
Realtime Error Event
-
class RealtimeErrorEvent: …Returned when an error occurs, which could be a client problem or a server problem. Most errors are recoverable and the session will stay open, we recommend to implementors to monitor and log error messages by default.
-
error: RealtimeErrorDetails of the error.
-
message: strA human-readable error message.
-
type: strThe type of error (e.g., "invalid_request_error", "server_error").
-
code: Optional[str]Error code, if any.
-
event_id: Optional[str]The event_id of the client event that caused the error, if applicable.
-
param: Optional[str]Parameter related to the error, if any.
-
-
event_id: strThe unique ID of the server event.
-
type: Literal["error"]The event type, must be
error."error"
-
Realtime Function Tool
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
Realtime Mcp Approval Request
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
Realtime Mcp Approval Response
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
Realtime Mcp List Tools
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
Realtime Mcp Protocol Error
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
Realtime Mcp Tool Call
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
Realtime Mcp Tool Execution Error
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
Realtime Mcphttp Error
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
Realtime Reasoning
-
class RealtimeReasoning: …Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
Realtime Reasoning Effort
-
Literal["minimal", "low", "medium", 2 more]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
Realtime Response
-
class RealtimeResponse: …The response resource.
-
id: Optional[str]The unique ID of the response, will look like
resp_1234. -
audio: Optional[Audio]Configuration for audio output.
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
conversation_id: Optional[str]Which conversation the response is added to, determined by the
conversationfield in theresponse.createevent. Ifauto, the response will be added to the default conversation and the value ofconversation_idwill be an id likeconv_1234. Ifnone, the response will not be added to any conversation and the value ofconversation_idwill benull. If responses are being triggered automatically by VAD the response will be added to the default conversation -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls, that was used in this response.
-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
object: Optional[Literal["realtime.response"]]The object type, must be
realtime.response."realtime.response"
-
output: Optional[List[ConversationItem]]The list of output items generated by the response.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
status: Optional[Literal["completed", "cancelled", "failed", 2 more]]The final status of the response (
completed,cancelled,failed, orincomplete,in_progress).-
"completed" -
"cancelled" -
"failed" -
"incomplete" -
"in_progress"
-
-
status_details: Optional[RealtimeResponseStatus]Additional details about the status.
-
error: Optional[Error]A description of the error that caused the response to fail, populated when the
statusisfailed.-
code: Optional[str]Error code, if any.
-
type: Optional[str]The type of error.
-
-
reason: Optional[Literal["turn_detected", "client_cancelled", "max_output_tokens", "content_filter"]]The reason the Response did not complete. For a
cancelledResponse, one ofturn_detected(the server VAD detected a new start of speech) orclient_cancelled(the client sent a cancel event). For anincompleteResponse, one ofmax_output_tokensorcontent_filter(the server-side safety filter activated and cut off the response).-
"turn_detected" -
"client_cancelled" -
"max_output_tokens" -
"content_filter"
-
-
type: Optional[Literal["completed", "cancelled", "incomplete", "failed"]]The type of error that caused the response to fail, corresponding with the
statusfield (completed,cancelled,incomplete,failed).-
"completed" -
"cancelled" -
"incomplete" -
"failed"
-
-
-
usage: Optional[RealtimeResponseUsage]Usage statistics for the Response, this will correspond to billing. A Realtime API session will maintain a conversation context and append new Items to the Conversation, thus output from previous turns (text and audio tokens) will become the input for later turns.
-
input_token_details: Optional[RealtimeResponseUsageInputTokenDetails]Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
-
input_tokens: Optional[int]The number of input tokens used in the Response, including text and audio tokens.
-
output_token_details: Optional[RealtimeResponseUsageOutputTokenDetails]Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
-
output_tokens: Optional[int]The number of output tokens sent in the Response, including text and audio tokens.
-
total_tokens: Optional[int]The total number of tokens in the Response including input and output text and audio tokens.
-
-
Realtime Response Create Audio Output
-
class RealtimeResponseCreateAudioOutput: …Configuration for audio input and output.
-
output: Optional[Output]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[OutputVoice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class OutputVoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
Realtime Response Create Mcp Tool
-
class RealtimeResponseCreateMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[AllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class AllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[RequireApproval]Specify which of the MCP server's tools require approval.
-
class RequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[RequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[RequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
Realtime Response Create Params
-
class RealtimeResponseCreateParams: …Create a new Realtime response with these parameters
-
audio: Optional[RealtimeResponseCreateAudioOutput]Configuration for audio input and output.
-
output: Optional[Output]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[OutputVoice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class OutputVoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
conversation: Optional[Union[str, Literal["auto", "none"], null]]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
str -
Literal["auto", "none"]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
"auto" -
"none"
-
-
-
input: Optional[List[ConversationItem]]Input items to include in the prompt for the model. Using this field creates a new context for this Response instead of using the default conversation. An empty array
[]will clear the context for this Response. Note that this can include references to items that previously appeared in the session using their id.-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior. Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class RealtimeResponseCreateMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[AllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class AllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[RequireApproval]Specify which of the MCP server's tools require approval.
-
class RequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[RequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[RequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
Realtime Response Status
-
class RealtimeResponseStatus: …Additional details about the status.
-
error: Optional[Error]A description of the error that caused the response to fail, populated when the
statusisfailed.-
code: Optional[str]Error code, if any.
-
type: Optional[str]The type of error.
-
-
reason: Optional[Literal["turn_detected", "client_cancelled", "max_output_tokens", "content_filter"]]The reason the Response did not complete. For a
cancelledResponse, one ofturn_detected(the server VAD detected a new start of speech) orclient_cancelled(the client sent a cancel event). For anincompleteResponse, one ofmax_output_tokensorcontent_filter(the server-side safety filter activated and cut off the response).-
"turn_detected" -
"client_cancelled" -
"max_output_tokens" -
"content_filter"
-
-
type: Optional[Literal["completed", "cancelled", "incomplete", "failed"]]The type of error that caused the response to fail, corresponding with the
statusfield (completed,cancelled,incomplete,failed).-
"completed" -
"cancelled" -
"incomplete" -
"failed"
-
-
Realtime Response Usage
-
class RealtimeResponseUsage: …Usage statistics for the Response, this will correspond to billing. A Realtime API session will maintain a conversation context and append new Items to the Conversation, thus output from previous turns (text and audio tokens) will become the input for later turns.
-
input_token_details: Optional[RealtimeResponseUsageInputTokenDetails]Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
-
input_tokens: Optional[int]The number of input tokens used in the Response, including text and audio tokens.
-
output_token_details: Optional[RealtimeResponseUsageOutputTokenDetails]Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
-
output_tokens: Optional[int]The number of output tokens sent in the Response, including text and audio tokens.
-
total_tokens: Optional[int]The total number of tokens in the Response including input and output text and audio tokens.
-
Realtime Response Usage Input Token Details
-
class RealtimeResponseUsageInputTokenDetails: …Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
Realtime Response Usage Output Token Details
-
class RealtimeResponseUsageOutputTokenDetails: …Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
Realtime Server Event
-
RealtimeServerEventA realtime server event.
-
class ConversationCreatedEvent: …Returned when a conversation is created. Emitted right after session creation.
-
conversation: ConversationThe conversation resource.
-
id: Optional[str]The unique ID of the conversation.
-
object: Optional[Literal["realtime.conversation"]]The object type, must be
realtime.conversation."realtime.conversation"
-
-
event_id: strThe unique ID of the server event.
-
type: Literal["conversation.created"]The event type, must be
conversation.created."conversation.created"
-
-
class ConversationItemCreatedEvent: …Returned when a conversation item is created. There are several scenarios that produce this event:
-
The server is generating a Response, which if successful will produce either one or two Items, which will be of type
message(roleassistant) or typefunction_call. -
The input audio buffer has been committed, either by the client or the server (in
server_vadmode). The server will take the content of the input audio buffer and add it to a new user message Item. -
The client has sent a
conversation.item.createevent to add a new Item to the Conversation. -
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
type: Literal["conversation.item.created"]The event type, must be
conversation.item.created."conversation.item.created"
-
previous_item_id: Optional[str]The ID of the preceding item in the Conversation context, allows the client to understand the order of the conversation. Can be
nullif the item has no predecessor.
-
-
class ConversationItemDeletedEvent: …Returned when an item in the conversation is deleted by the client with a
conversation.item.deleteevent. This event is used to synchronize the server's understanding of the conversation history with the client's view.-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item that was deleted.
-
type: Literal["conversation.item.deleted"]The event type, must be
conversation.item.deleted."conversation.item.deleted"
-
-
class ConversationItemInputAudioTranscriptionCompletedEvent: …This event is the output of audio transcription for user audio written to the user audio buffer. Transcription begins when the input audio buffer is committed by the client or server (when VAD is enabled). Transcription runs asynchronously with Response creation, so this event may come before or after the Response events.
Realtime API models accept audio natively, and thus input transcription is a separate process run on a separate ASR (Automatic Speech Recognition) model. The transcript may diverge somewhat from the model's interpretation, and should be treated as a rough guide.
-
content_index: intThe index of the content part containing the audio.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the audio that is being transcribed.
-
transcript: strThe transcribed text.
-
type: Literal["conversation.item.input_audio_transcription.completed"]The event type, must be
conversation.item.input_audio_transcription.completed."conversation.item.input_audio_transcription.completed"
-
usage: UsageUsage statistics for the transcription, this is billed according to the ASR model's pricing rather than the realtime model's pricing.
-
class UsageTranscriptTextUsageTokens: …Usage statistics for models billed by token usage.
-
input_tokens: intNumber of input tokens billed for this request.
-
output_tokens: intNumber of output tokens generated.
-
total_tokens: intTotal number of tokens used (input + output).
-
type: Literal["tokens"]The type of the usage object. Always
tokensfor this variant."tokens"
-
input_token_details: Optional[UsageTranscriptTextUsageTokensInputTokenDetails]Details about the input tokens billed for this request.
-
audio_tokens: Optional[int]Number of audio tokens billed for this request.
-
text_tokens: Optional[int]Number of text tokens billed for this request.
-
-
-
class UsageTranscriptTextUsageDuration: …Usage statistics for models billed by audio input duration.
-
seconds: floatDuration of the input audio in seconds.
-
type: Literal["duration"]The type of the usage object. Always
durationfor this variant."duration"
-
-
-
logprobs: Optional[List[LogProbProperties]]The log probabilities of the transcription.
-
token: strThe token that was used to generate the log probability.
-
bytes: List[int]The bytes that were used to generate the log probability.
-
logprob: floatThe log probability of the token.
-
-
-
class ConversationItemInputAudioTranscriptionDeltaEvent: …Returned when the text value of an input audio transcription content part is updated with incremental transcription results.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the audio that is being transcribed.
-
type: Literal["conversation.item.input_audio_transcription.delta"]The event type, must be
conversation.item.input_audio_transcription.delta."conversation.item.input_audio_transcription.delta"
-
content_index: Optional[int]The index of the content part in the item's content array.
-
delta: Optional[str]The text delta.
-
logprobs: Optional[List[LogProbProperties]]The log probabilities of the transcription. These can be enabled by configurating the session with
"include": ["item.input_audio_transcription.logprobs"]. Each entry in the array corresponds a log probability of which token would be selected for this chunk of transcription. This can help to identify if it was possible there were multiple valid options for a given chunk of transcription.-
token: strThe token that was used to generate the log probability.
-
bytes: List[int]The bytes that were used to generate the log probability.
-
logprob: floatThe log probability of the token.
-
-
-
class ConversationItemInputAudioTranscriptionFailedEvent: …Returned when input audio transcription is configured, and a transcription request for a user message failed. These events are separate from other
errorevents so that the client can identify the related Item.-
content_index: intThe index of the content part containing the audio.
-
error: ErrorDetails of the transcription error.
-
code: Optional[str]Error code, if any.
-
message: Optional[str]A human-readable error message.
-
param: Optional[str]Parameter related to the error, if any.
-
type: Optional[str]The type of error.
-
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item.
-
type: Literal["conversation.item.input_audio_transcription.failed"]The event type, must be
conversation.item.input_audio_transcription.failed."conversation.item.input_audio_transcription.failed"
-
-
class ConversationItemRetrieved: …Returned when a conversation item is retrieved with
conversation.item.retrieve. This is provided as a way to fetch the server's representation of an item, for example to get access to the post-processed audio data after noise cancellation and VAD. It includes the full content of the Item, including audio data.-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
type: Literal["conversation.item.retrieved"]The event type, must be
conversation.item.retrieved."conversation.item.retrieved"
-
-
class ConversationItemTruncatedEvent: …Returned when an earlier assistant audio message item is truncated by the client with a
conversation.item.truncateevent. This event is used to synchronize the server's understanding of the audio with the client's playback.This action will truncate the audio and remove the server-side text transcript to ensure there is no text in the context that hasn't been heard by the user.
-
audio_end_ms: intThe duration up to which the audio was truncated, in milliseconds.
-
content_index: intThe index of the content part that was truncated.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the assistant message item that was truncated.
-
type: Literal["conversation.item.truncated"]The event type, must be
conversation.item.truncated."conversation.item.truncated"
-
-
class RealtimeErrorEvent: …Returned when an error occurs, which could be a client problem or a server problem. Most errors are recoverable and the session will stay open, we recommend to implementors to monitor and log error messages by default.
-
error: RealtimeErrorDetails of the error.
-
message: strA human-readable error message.
-
type: strThe type of error (e.g., "invalid_request_error", "server_error").
-
code: Optional[str]Error code, if any.
-
event_id: Optional[str]The event_id of the client event that caused the error, if applicable.
-
param: Optional[str]Parameter related to the error, if any.
-
-
event_id: strThe unique ID of the server event.
-
type: Literal["error"]The event type, must be
error."error"
-
-
class InputAudioBufferClearedEvent: …Returned when the input audio buffer is cleared by the client with a
input_audio_buffer.clearevent.-
event_id: strThe unique ID of the server event.
-
type: Literal["input_audio_buffer.cleared"]The event type, must be
input_audio_buffer.cleared."input_audio_buffer.cleared"
-
-
class InputAudioBufferCommittedEvent: …Returned when an input audio buffer is committed, either by the client or automatically in server VAD mode. The
item_idproperty is the ID of the user message item that will be created, thus aconversation.item.createdevent will also be sent to the client.-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created.
-
type: Literal["input_audio_buffer.committed"]The event type, must be
input_audio_buffer.committed."input_audio_buffer.committed"
-
previous_item_id: Optional[str]The ID of the preceding item after which the new item will be inserted. Can be
nullif the item has no predecessor.
-
-
class InputAudioBufferDtmfEventReceivedEvent: …SIP Only: Returned when an DTMF event is received. A DTMF event is a message that represents a telephone keypad press (0–9, *, #, A–D). The
eventproperty is the keypad that the user press. Thereceived_atis the UTC Unix Timestamp that the server received the event.-
event: strThe telephone keypad that was pressed by the user.
-
received_at: intUTC Unix Timestamp when DTMF Event was received by server.
-
type: Literal["input_audio_buffer.dtmf_event_received"]The event type, must be
input_audio_buffer.dtmf_event_received."input_audio_buffer.dtmf_event_received"
-
-
class InputAudioBufferSpeechStartedEvent: …Sent by the server when in
server_vadmode to indicate that speech has been detected in the audio buffer. This can happen any time audio is added to the buffer (unless speech is already detected). The client may want to use this event to interrupt audio playback or provide visual feedback to the user.The client should expect to receive a
input_audio_buffer.speech_stoppedevent when speech stops. Theitem_idproperty is the ID of the user message item that will be created when speech stops and will also be included in theinput_audio_buffer.speech_stoppedevent (unless the client manually commits the audio buffer during VAD activation).-
audio_start_ms: intMilliseconds from the start of all audio written to the buffer during the session when speech was first detected. This will correspond to the beginning of audio sent to the model, and thus includes the
prefix_padding_msconfigured in the Session. -
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created when speech stops.
-
type: Literal["input_audio_buffer.speech_started"]The event type, must be
input_audio_buffer.speech_started."input_audio_buffer.speech_started"
-
-
class InputAudioBufferSpeechStoppedEvent: …Returned in
server_vadmode when the server detects the end of speech in the audio buffer. The server will also send anconversation.item.createdevent with the user message item that is created from the audio buffer.-
audio_end_ms: intMilliseconds since the session started when speech stopped. This will correspond to the end of audio sent to the model, and thus includes the
min_silence_duration_msconfigured in the Session. -
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the user message item that will be created.
-
type: Literal["input_audio_buffer.speech_stopped"]The event type, must be
input_audio_buffer.speech_stopped."input_audio_buffer.speech_stopped"
-
-
class RateLimitsUpdatedEvent: …Emitted at the beginning of a Response to indicate the updated rate limits. When a Response is created some tokens will be "reserved" for the output tokens, the rate limits shown here reflect that reservation, which is then adjusted accordingly once the Response is completed.
-
event_id: strThe unique ID of the server event.
-
rate_limits: List[RateLimit]List of rate limit information.
-
limit: Optional[int]The maximum allowed value for the rate limit.
-
name: Optional[Literal["requests", "tokens"]]The name of the rate limit (
requests,tokens).-
"requests" -
"tokens"
-
-
remaining: Optional[int]The remaining value before the limit is reached.
-
reset_seconds: Optional[float]Seconds until the rate limit resets.
-
-
type: Literal["rate_limits.updated"]The event type, must be
rate_limits.updated."rate_limits.updated"
-
-
class ResponseAudioDeltaEvent: …Returned when the model-generated audio is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strBase64-encoded audio data delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio.delta"]The event type, must be
response.output_audio.delta."response.output_audio.delta"
-
-
class ResponseAudioDoneEvent: …Returned when the model-generated audio is done. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio.done"]The event type, must be
response.output_audio.done."response.output_audio.done"
-
-
class ResponseAudioTranscriptDeltaEvent: …Returned when the model-generated transcription of audio output is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strThe transcript delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio_transcript.delta"]The event type, must be
response.output_audio_transcript.delta."response.output_audio_transcript.delta"
-
-
class ResponseAudioTranscriptDoneEvent: …Returned when the model-generated transcription of audio output is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
transcript: strThe final transcript of the audio.
-
type: Literal["response.output_audio_transcript.done"]The event type, must be
response.output_audio_transcript.done."response.output_audio_transcript.done"
-
-
class ResponseContentPartAddedEvent: …Returned when a new content part is added to an assistant message item during response generation.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item to which the content part was added.
-
output_index: intThe index of the output item in the response.
-
part: PartThe content part that was added.
-
audio: Optional[str]Base64-encoded audio data (if type is "audio").
-
text: Optional[str]The text content (if type is "text").
-
transcript: Optional[str]The transcript of the audio (if type is "audio").
-
type: Optional[Literal["text", "audio"]]The content type ("text", "audio").
-
"text" -
"audio"
-
-
-
response_id: strThe ID of the response.
-
type: Literal["response.content_part.added"]The event type, must be
response.content_part.added."response.content_part.added"
-
-
class ResponseContentPartDoneEvent: …Returned when a content part is done streaming in an assistant message item. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
part: PartThe content part that is done.
-
audio: Optional[str]Base64-encoded audio data (if type is "audio").
-
text: Optional[str]The text content (if type is "text").
-
transcript: Optional[str]The transcript of the audio (if type is "audio").
-
type: Optional[Literal["text", "audio"]]The content type ("text", "audio").
-
"text" -
"audio"
-
-
-
response_id: strThe ID of the response.
-
type: Literal["response.content_part.done"]The event type, must be
response.content_part.done."response.content_part.done"
-
-
class ResponseCreatedEvent: …Returned when a new Response is created. The first event of response creation, where the response is in an initial state of
in_progress.-
event_id: strThe unique ID of the server event.
-
response: RealtimeResponseThe response resource.
-
id: Optional[str]The unique ID of the response, will look like
resp_1234. -
audio: Optional[Audio]Configuration for audio output.
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
conversation_id: Optional[str]Which conversation the response is added to, determined by the
conversationfield in theresponse.createevent. Ifauto, the response will be added to the default conversation and the value ofconversation_idwill be an id likeconv_1234. Ifnone, the response will not be added to any conversation and the value ofconversation_idwill benull. If responses are being triggered automatically by VAD the response will be added to the default conversation -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls, that was used in this response.
-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
object: Optional[Literal["realtime.response"]]The object type, must be
realtime.response."realtime.response"
-
output: Optional[List[ConversationItem]]The list of output items generated by the response.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
status: Optional[Literal["completed", "cancelled", "failed", 2 more]]The final status of the response (
completed,cancelled,failed, orincomplete,in_progress).-
"completed" -
"cancelled" -
"failed" -
"incomplete" -
"in_progress"
-
-
status_details: Optional[RealtimeResponseStatus]Additional details about the status.
-
error: Optional[Error]A description of the error that caused the response to fail, populated when the
statusisfailed.-
code: Optional[str]Error code, if any.
-
type: Optional[str]The type of error.
-
-
reason: Optional[Literal["turn_detected", "client_cancelled", "max_output_tokens", "content_filter"]]The reason the Response did not complete. For a
cancelledResponse, one ofturn_detected(the server VAD detected a new start of speech) orclient_cancelled(the client sent a cancel event). For anincompleteResponse, one ofmax_output_tokensorcontent_filter(the server-side safety filter activated and cut off the response).-
"turn_detected" -
"client_cancelled" -
"max_output_tokens" -
"content_filter"
-
-
type: Optional[Literal["completed", "cancelled", "incomplete", "failed"]]The type of error that caused the response to fail, corresponding with the
statusfield (completed,cancelled,incomplete,failed).-
"completed" -
"cancelled" -
"incomplete" -
"failed"
-
-
-
usage: Optional[RealtimeResponseUsage]Usage statistics for the Response, this will correspond to billing. A Realtime API session will maintain a conversation context and append new Items to the Conversation, thus output from previous turns (text and audio tokens) will become the input for later turns.
-
input_token_details: Optional[RealtimeResponseUsageInputTokenDetails]Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
-
input_tokens: Optional[int]The number of input tokens used in the Response, including text and audio tokens.
-
output_token_details: Optional[RealtimeResponseUsageOutputTokenDetails]Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
-
output_tokens: Optional[int]The number of output tokens sent in the Response, including text and audio tokens.
-
total_tokens: Optional[int]The total number of tokens in the Response including input and output text and audio tokens.
-
-
-
type: Literal["response.created"]The event type, must be
response.created."response.created"
-
-
class ResponseDoneEvent: …Returned when a Response is done streaming. Always emitted, no matter the final state. The Response object included in the
response.doneevent will include all output Items in the Response but will omit the raw audio data.Clients should check the
statusfield of the Response to determine if it was successful (completed) or if there was another outcome:cancelled,failed, orincomplete.A response will contain all output items that were generated during the response, excluding any audio content.
-
event_id: strThe unique ID of the server event.
-
response: RealtimeResponseThe response resource.
-
type: Literal["response.done"]The event type, must be
response.done."response.done"
-
-
class ResponseFunctionCallArgumentsDeltaEvent: …Returned when the model-generated function call arguments are updated.
-
call_id: strThe ID of the function call.
-
delta: strThe arguments delta as a JSON string.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the function call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.function_call_arguments.delta"]The event type, must be
response.function_call_arguments.delta."response.function_call_arguments.delta"
-
-
class ResponseFunctionCallArgumentsDoneEvent: …Returned when the model-generated function call arguments are done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
arguments: strThe final arguments as a JSON string.
-
call_id: strThe ID of the function call.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the function call item.
-
name: strThe name of the function that was called.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.function_call_arguments.done"]The event type, must be
response.function_call_arguments.done."response.function_call_arguments.done"
-
-
class ResponseOutputItemAddedEvent: …Returned when a new Item is created during Response generation.
-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
output_index: intThe index of the output item in the Response.
-
response_id: strThe ID of the Response to which the item belongs.
-
type: Literal["response.output_item.added"]The event type, must be
response.output_item.added."response.output_item.added"
-
-
class ResponseOutputItemDoneEvent: …Returned when an Item is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
output_index: intThe index of the output item in the Response.
-
response_id: strThe ID of the Response to which the item belongs.
-
type: Literal["response.output_item.done"]The event type, must be
response.output_item.done."response.output_item.done"
-
-
class ResponseTextDeltaEvent: …Returned when the text value of an "output_text" content part is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strThe text delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_text.delta"]The event type, must be
response.output_text.delta."response.output_text.delta"
-
-
class ResponseTextDoneEvent: …Returned when the text value of an "output_text" content part is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
text: strThe final text content.
-
type: Literal["response.output_text.done"]The event type, must be
response.output_text.done."response.output_text.done"
-
-
class SessionCreatedEvent: …Returned when a Session is created. Emitted automatically when a new connection is established as the first server event. This event will contain the default Session configuration.
-
event_id: strThe unique ID of the server event.
-
session: SessionThe session configuration.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
-
type: Literal["session.created"]The event type, must be
session.created."session.created"
-
-
class SessionUpdatedEvent: …Returned when a session is updated with a
session.updateevent, unless there is an error.-
event_id: strThe unique ID of the server event.
-
session: SessionThe session configuration.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
-
type: Literal["session.updated"]The event type, must be
session.updated."session.updated"
-
-
class OutputAudioBufferStarted: …WebRTC/SIP Only: Emitted when the server begins streaming audio to the client. This event is emitted after an audio content part has been added (
response.content_part.added) to the response. Learn more.-
event_id: strThe unique ID of the server event.
-
response_id: strThe unique ID of the response that produced the audio.
-
type: Literal["output_audio_buffer.started"]The event type, must be
output_audio_buffer.started."output_audio_buffer.started"
-
-
class OutputAudioBufferStopped: …WebRTC/SIP Only: Emitted when the output audio buffer has been completely drained on the server, and no more audio is forthcoming. This event is emitted after the full response data has been sent to the client (
response.done). Learn more.-
event_id: strThe unique ID of the server event.
-
response_id: strThe unique ID of the response that produced the audio.
-
type: Literal["output_audio_buffer.stopped"]The event type, must be
output_audio_buffer.stopped."output_audio_buffer.stopped"
-
-
class OutputAudioBufferCleared: …WebRTC/SIP Only: Emitted when the output audio buffer is cleared. This happens either in VAD mode when the user has interrupted (
input_audio_buffer.speech_started), or when the client has emitted theoutput_audio_buffer.clearevent to manually cut off the current audio response. Learn more.-
event_id: strThe unique ID of the server event.
-
response_id: strThe unique ID of the response that produced the audio.
-
type: Literal["output_audio_buffer.cleared"]The event type, must be
output_audio_buffer.cleared."output_audio_buffer.cleared"
-
-
class ConversationItemAdded: …Sent by the server when an Item is added to the default Conversation. This can happen in several cases:
- When the client sends a
conversation.item.createevent. - When the input audio buffer is committed. In this case the item will be a user message containing the audio from the buffer.
- When the model is generating a Response. In this case the
conversation.item.addedevent will be sent when the model starts generating a specific Item, and thus it will not yet have any content (andstatuswill bein_progress).
The event will include the full content of the Item (except when model is generating a Response) except for audio data, which can be retrieved separately with a
conversation.item.retrieveevent if necessary.-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
type: Literal["conversation.item.added"]The event type, must be
conversation.item.added."conversation.item.added"
-
previous_item_id: Optional[str]The ID of the item that precedes this one, if any. This is used to maintain ordering when items are inserted.
- When the client sends a
-
class ConversationItemDone: …Returned when a conversation item is finalized.
The event will include the full content of the Item except for audio data, which can be retrieved separately with a
conversation.item.retrieveevent if needed.-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
type: Literal["conversation.item.done"]The event type, must be
conversation.item.done."conversation.item.done"
-
previous_item_id: Optional[str]The ID of the item that precedes this one, if any. This is used to maintain ordering when items are inserted.
-
-
class InputAudioBufferTimeoutTriggered: …Returned when the Server VAD timeout is triggered for the input audio buffer. This is configured with
idle_timeout_msin theturn_detectionsettings of the session, and it indicates that there hasn't been any speech detected for the configured duration.The
audio_start_msandaudio_end_msfields indicate the segment of audio after the last model response up to the triggering time, as an offset from the beginning of audio written to the input audio buffer. This means it demarcates the segment of audio that was silent and the difference between the start and end values will roughly match the configured timeout.The empty audio will be committed to the conversation as an
input_audioitem (there will be ainput_audio_buffer.committedevent) and a model response will be generated. There may be speech that didn't trigger VAD but is still detected by the model, so the model may respond with something relevant to the conversation or a prompt to continue speaking.-
audio_end_ms: intMillisecond offset of audio written to the input audio buffer at the time the timeout was triggered.
-
audio_start_ms: intMillisecond offset of audio written to the input audio buffer that was after the playback time of the last model response.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item associated with this segment.
-
type: Literal["input_audio_buffer.timeout_triggered"]The event type, must be
input_audio_buffer.timeout_triggered."input_audio_buffer.timeout_triggered"
-
-
class ConversationItemInputAudioTranscriptionSegment: …Returned when an input audio transcription segment is identified for an item.
-
id: strThe segment identifier.
-
content_index: intThe index of the input audio content part within the item.
-
end: floatEnd time of the segment in seconds.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item containing the input audio content.
-
speaker: strThe detected speaker label for this segment.
-
start: floatStart time of the segment in seconds.
-
text: strThe text for this segment.
-
type: Literal["conversation.item.input_audio_transcription.segment"]The event type, must be
conversation.item.input_audio_transcription.segment."conversation.item.input_audio_transcription.segment"
-
-
class McpListToolsInProgress: …Returned when listing MCP tools is in progress for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.in_progress"]The event type, must be
mcp_list_tools.in_progress."mcp_list_tools.in_progress"
-
-
class McpListToolsCompleted: …Returned when listing MCP tools has completed for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.completed"]The event type, must be
mcp_list_tools.completed."mcp_list_tools.completed"
-
-
class McpListToolsFailed: …Returned when listing MCP tools has failed for an item.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP list tools item.
-
type: Literal["mcp_list_tools.failed"]The event type, must be
mcp_list_tools.failed."mcp_list_tools.failed"
-
-
class ResponseMcpCallArgumentsDelta: …Returned when MCP tool call arguments are updated during response generation.
-
delta: strThe JSON-encoded arguments delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.mcp_call_arguments.delta"]The event type, must be
response.mcp_call_arguments.delta."response.mcp_call_arguments.delta"
-
obfuscation: Optional[str]If present, indicates the delta text was obfuscated.
-
-
class ResponseMcpCallArgumentsDone: …Returned when MCP tool call arguments are finalized during response generation.
-
arguments: strThe final JSON-encoded arguments string.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.mcp_call_arguments.done"]The event type, must be
response.mcp_call_arguments.done."response.mcp_call_arguments.done"
-
-
class ResponseMcpCallInProgress: …Returned when an MCP tool call has started and is in progress.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.in_progress"]The event type, must be
response.mcp_call.in_progress."response.mcp_call.in_progress"
-
-
class ResponseMcpCallCompleted: …Returned when an MCP tool call has completed successfully.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.completed"]The event type, must be
response.mcp_call.completed."response.mcp_call.completed"
-
-
class ResponseMcpCallFailed: …Returned when an MCP tool call has failed.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.failed"]The event type, must be
response.mcp_call.failed."response.mcp_call.failed"
-
-
Realtime Session
-
class RealtimeSession: …Realtime session object for the beta interface.
-
id: Optional[str]Unique identifier for the session that looks like
sess_1234567890abcdef. -
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
-
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription. -
"item.input_audio_transcription.logprobs"
-
-
input_audio_format: Optional[Literal["pcm16", "g711_ulaw", "g711_alaw"]]The format of input audio. Options are
pcm16,g711_ulaw, org711_alaw. Forpcm16, input audio must be 16-bit PCM at a 24kHz sample rate, single channel (mono), and little-endian byte order.-
"pcm16" -
"g711_ulaw" -
"g711_alaw"
-
-
input_audio_noise_reduction: Optional[InputAudioNoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
input_audio_transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_response_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. To disable audio, set this to ["text"].
-
"text" -
"audio"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2025-08-28", 13 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2025-08-28", 13 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
object: Optional[Literal["realtime.session"]]The object type. Always
realtime.session."realtime.session"
-
output_audio_format: Optional[Literal["pcm16", "g711_ulaw", "g711_alaw"]]The format of output audio. Options are
pcm16,g711_ulaw, org711_alaw. Forpcm16, output audio is sampled at a rate of 24kHz.-
"pcm16" -
"g711_ulaw" -
"g711_alaw"
-
-
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
speed: Optional[float]The speed of the model's spoken response. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
-
temperature: Optional[float]Sampling temperature for the model, limited to [0.6, 1.2]. For audio models a temperature of 0.8 is highly recommended for best performance.
-
tool_choice: Optional[str]How the model chooses tools. Options are
auto,none,required, or specify a function. -
tools: Optional[List[RealtimeFunctionTool]]Tools (functions) available to the model.
-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
tracing: Optional[Tracing]Configuration options for tracing. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Default tracing mode for the session.
"auto"
-
class TracingTracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the traces dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the traces dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the traces dashboard.
-
-
-
turn_detection: Optional[TurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class TurnDetectionServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class TurnDetectionSemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer, andverse.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer, andverse.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
Realtime Session Create Request
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
Realtime Tool Choice Config
-
RealtimeToolChoiceConfigHow the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
Realtime Tools Config
-
List[RealtimeToolsConfigUnion]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
Realtime Tools Config Union
-
RealtimeToolsConfigUnionGive the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
Realtime Tracing Config
-
Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
Realtime Transcription Session Audio
-
class RealtimeTranscriptionSessionAudio: …Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
Realtime Transcription Session Audio Input
-
class RealtimeTranscriptionSessionAudioInput: …-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
Realtime Transcription Session Audio Input Turn Detection
-
Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
Realtime Transcription Session Create Request
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
Realtime Translation Client Event
-
RealtimeTranslationClientEventA Realtime translation client event.
-
class RealtimeTranslationSessionUpdateEvent: …Send this event to update the translation session configuration. Translation sessions support updates to
audio.output.language,audio.input.transcription, andaudio.input.noise_reduction.-
session: RealtimeTranslationSessionUpdateRequestTranslation session fields to update. The session
typeandmodelare set at creation and cannot be changed withsession.update.-
audio: Optional[Audio]Configuration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction. Set to
nullto disable it.-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model to use for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
-
type: Literal["session.update"]The event type, must be
session.update."session.update"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class RealtimeTranslationInputAudioBufferAppendEvent: …Send this event to append audio bytes to the translation session input audio buffer.
WebSocket translation sessions accept base64-encoded 24 kHz PCM16 mono little-endian raw audio bytes. Unsupported websocket audio formats return a validation error because lower-quality audio materially degrades translation quality.
Translation consumes 200 ms engine frames. For best realtime behavior, append audio in 200 ms chunks. If a chunk is shorter, the server buffers it until it has enough audio for one frame. If a chunk is longer, the server splits it into 200 ms frames and enqueues them back-to-back.
Keep appending silence while the session is active. If a client stops sending audio and later resumes, model time treats the resumed audio as contiguous with the previous audio rather than as a real-world pause.
-
audio: strBase64-encoded 24 kHz PCM16 mono audio bytes.
-
type: Literal["session.input_audio_buffer.append"]The event type, must be
session.input_audio_buffer.append."session.input_audio_buffer.append"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
class RealtimeTranslationSessionCloseEvent: …Gracefully close the realtime translation session. The server flushes pending input audio and emits any remaining translated output before closing the session.
-
type: Literal["session.close"]The event type, must be
session.close."session.close"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
-
Realtime Translation Client Secret Create Request
-
class RealtimeTranslationClientSecretCreateRequest: …Create a translation session and client secret for the Realtime API.
-
session: RealtimeTranslationSessionCreateRequestRealtime translation session configuration. Translation sessions stream source audio in and translated audio plus transcript deltas out continuously.
-
model: strThe Realtime translation model used for this session.
-
audio: Optional[Audio]Configuration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction. Set to
nullto disable it.-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model to use for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
-
expires_after: Optional[ExpiresAfter]Configuration for the client secret expiration. Expiration refers to the time after which a client secret will no longer be valid for creating sessions. The session itself may continue after that time once started. A secret can be used to create multiple sessions until it expires.
-
anchor: Optional[Literal["created_at"]]The anchor point for the client secret expiration, meaning that
secondswill be added to thecreated_attime of the client secret to produce an expiration timestamp. Onlycreated_atis currently supported."created_at"
-
seconds: Optional[int]The number of seconds from the anchor point to the expiration. Select a value between
10and7200(2 hours). This default to 600 seconds (10 minutes) if not specified.
-
-
Realtime Translation Client Secret Create Response
-
class RealtimeTranslationClientSecretCreateResponse: …Response from creating a translation session and client secret for the Realtime API.
-
expires_at: intExpiration timestamp for the client secret, in seconds since epoch.
-
session: RealtimeTranslationSessionA Realtime translation session. Translation sessions continuously translate input audio into the configured output language.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
audio: AudioConfiguration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction.
-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model used for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
expires_at: intExpiration timestamp for the session, in seconds since epoch.
-
model: strThe Realtime translation model used for this session. This field is set at session creation and cannot be changed with
session.update. -
type: Literal["translation"]The session type. Always
translationfor Realtime translation sessions."translation"
-
-
value: strThe generated client secret value.
-
Realtime Translation Input Audio Buffer Append Event
-
class RealtimeTranslationInputAudioBufferAppendEvent: …Send this event to append audio bytes to the translation session input audio buffer.
WebSocket translation sessions accept base64-encoded 24 kHz PCM16 mono little-endian raw audio bytes. Unsupported websocket audio formats return a validation error because lower-quality audio materially degrades translation quality.
Translation consumes 200 ms engine frames. For best realtime behavior, append audio in 200 ms chunks. If a chunk is shorter, the server buffers it until it has enough audio for one frame. If a chunk is longer, the server splits it into 200 ms frames and enqueues them back-to-back.
Keep appending silence while the session is active. If a client stops sending audio and later resumes, model time treats the resumed audio as contiguous with the previous audio rather than as a real-world pause.
-
audio: strBase64-encoded 24 kHz PCM16 mono audio bytes.
-
type: Literal["session.input_audio_buffer.append"]The event type, must be
session.input_audio_buffer.append."session.input_audio_buffer.append"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Realtime Translation Input Transcript Delta Event
-
class RealtimeTranslationInputTranscriptDeltaEvent: …Returned when optional source-language transcript text is available. This event is emitted only when
audio.input.transcriptionis configured.Transcript deltas are append-only text fragments. Clients should not insert unconditional spaces between deltas.
-
delta: strAppend-only source-language transcript text.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.input_transcript.delta"]The event type, must be
session.input_transcript.delta."session.input_transcript.delta"
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. It advances in 200 ms increments, but multiple transcript deltas may share the same
elapsed_ms. Treat it as alignment metadata, not a unique transcript-delta identifier.
-
Realtime Translation Output Audio Delta Event
-
class RealtimeTranslationOutputAudioDeltaEvent: …Returned when translated output audio is available. Output audio deltas are 200 ms frames of PCM16 audio.
-
delta: strBase64-encoded translated audio data.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.output_audio.delta"]The event type, must be
session.output_audio.delta."session.output_audio.delta"
-
channels: Optional[int]Number of audio channels.
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. Treat
elapsed_msas alignment metadata, not a unique event identifier. -
format: Optional[Literal["pcm16"]]Audio encoding for
delta."pcm16"
-
sample_rate: Optional[int]Sample rate of the audio delta.
-
Realtime Translation Output Transcript Delta Event
-
class RealtimeTranslationOutputTranscriptDeltaEvent: …Returned when translated transcript text is available.
Transcript deltas are append-only text fragments. Clients should not insert unconditional spaces between deltas.
-
delta: strAppend-only transcript text for the translated output audio.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.output_transcript.delta"]The event type, must be
session.output_transcript.delta."session.output_transcript.delta"
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. It advances in 200 ms increments, but multiple transcript deltas may share the same
elapsed_ms. Treat it as alignment metadata, not a unique transcript-delta identifier.
-
Realtime Translation Server Event
-
RealtimeTranslationServerEventA Realtime translation server event.
-
class RealtimeErrorEvent: …Returned when an error occurs, which could be a client problem or a server problem. Most errors are recoverable and the session will stay open, we recommend to implementors to monitor and log error messages by default.
-
error: RealtimeErrorDetails of the error.
-
message: strA human-readable error message.
-
type: strThe type of error (e.g., "invalid_request_error", "server_error").
-
code: Optional[str]Error code, if any.
-
event_id: Optional[str]The event_id of the client event that caused the error, if applicable.
-
param: Optional[str]Parameter related to the error, if any.
-
-
event_id: strThe unique ID of the server event.
-
type: Literal["error"]The event type, must be
error."error"
-
-
class RealtimeTranslationSessionCreatedEvent: …Returned when a translation session is created. Emitted automatically when a new connection is established as the first server event. This event contains the default translation session configuration.
-
event_id: strThe unique ID of the server event.
-
session: RealtimeTranslationSessionThe translation session configuration.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
audio: AudioConfiguration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction.
-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model used for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
expires_at: intExpiration timestamp for the session, in seconds since epoch.
-
model: strThe Realtime translation model used for this session. This field is set at session creation and cannot be changed with
session.update. -
type: Literal["translation"]The session type. Always
translationfor Realtime translation sessions."translation"
-
-
type: Literal["session.created"]The event type, must be
session.created."session.created"
-
-
class RealtimeTranslationSessionUpdatedEvent: …Returned when a translation session is updated with a
session.updateevent, unless there is an error.-
event_id: strThe unique ID of the server event.
-
session: RealtimeTranslationSessionThe translation session configuration.
-
type: Literal["session.updated"]The event type, must be
session.updated."session.updated"
-
-
class RealtimeTranslationSessionClosedEvent: …Returned when a realtime translation session is closed.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.closed"]The event type, must be
session.closed."session.closed"
-
-
class RealtimeTranslationInputTranscriptDeltaEvent: …Returned when optional source-language transcript text is available. This event is emitted only when
audio.input.transcriptionis configured.Transcript deltas are append-only text fragments. Clients should not insert unconditional spaces between deltas.
-
delta: strAppend-only source-language transcript text.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.input_transcript.delta"]The event type, must be
session.input_transcript.delta."session.input_transcript.delta"
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. It advances in 200 ms increments, but multiple transcript deltas may share the same
elapsed_ms. Treat it as alignment metadata, not a unique transcript-delta identifier.
-
-
class RealtimeTranslationOutputTranscriptDeltaEvent: …Returned when translated transcript text is available.
Transcript deltas are append-only text fragments. Clients should not insert unconditional spaces between deltas.
-
delta: strAppend-only transcript text for the translated output audio.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.output_transcript.delta"]The event type, must be
session.output_transcript.delta."session.output_transcript.delta"
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. It advances in 200 ms increments, but multiple transcript deltas may share the same
elapsed_ms. Treat it as alignment metadata, not a unique transcript-delta identifier.
-
-
class RealtimeTranslationOutputAudioDeltaEvent: …Returned when translated output audio is available. Output audio deltas are 200 ms frames of PCM16 audio.
-
delta: strBase64-encoded translated audio data.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.output_audio.delta"]The event type, must be
session.output_audio.delta."session.output_audio.delta"
-
channels: Optional[int]Number of audio channels.
-
elapsed_ms: Optional[int]Timing metadata for stream alignment, derived from the translation frame when available. Treat
elapsed_msas alignment metadata, not a unique event identifier. -
format: Optional[Literal["pcm16"]]Audio encoding for
delta."pcm16"
-
sample_rate: Optional[int]Sample rate of the audio delta.
-
-
Realtime Translation Session
-
class RealtimeTranslationSession: …A Realtime translation session. Translation sessions continuously translate input audio into the configured output language.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
audio: AudioConfiguration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction.
-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model used for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
expires_at: intExpiration timestamp for the session, in seconds since epoch.
-
model: strThe Realtime translation model used for this session. This field is set at session creation and cannot be changed with
session.update. -
type: Literal["translation"]The session type. Always
translationfor Realtime translation sessions."translation"
-
Realtime Translation Session Close Event
-
class RealtimeTranslationSessionCloseEvent: …Gracefully close the realtime translation session. The server flushes pending input audio and emits any remaining translated output before closing the session.
-
type: Literal["session.close"]The event type, must be
session.close."session.close"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Realtime Translation Session Closed Event
-
class RealtimeTranslationSessionClosedEvent: …Returned when a realtime translation session is closed.
-
event_id: strThe unique ID of the server event.
-
type: Literal["session.closed"]The event type, must be
session.closed."session.closed"
-
Realtime Translation Session Create Request
-
class RealtimeTranslationSessionCreateRequest: …Realtime translation session configuration. Translation sessions stream source audio in and translated audio plus transcript deltas out continuously.
-
model: strThe Realtime translation model used for this session.
-
audio: Optional[Audio]Configuration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction. Set to
nullto disable it.-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model to use for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
Realtime Translation Session Created Event
-
class RealtimeTranslationSessionCreatedEvent: …Returned when a translation session is created. Emitted automatically when a new connection is established as the first server event. This event contains the default translation session configuration.
-
event_id: strThe unique ID of the server event.
-
session: RealtimeTranslationSessionThe translation session configuration.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
audio: AudioConfiguration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction.
-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model used for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
expires_at: intExpiration timestamp for the session, in seconds since epoch.
-
model: strThe Realtime translation model used for this session. This field is set at session creation and cannot be changed with
session.update. -
type: Literal["translation"]The session type. Always
translationfor Realtime translation sessions."translation"
-
-
type: Literal["session.created"]The event type, must be
session.created."session.created"
-
Realtime Translation Session Update Event
-
class RealtimeTranslationSessionUpdateEvent: …Send this event to update the translation session configuration. Translation sessions support updates to
audio.output.language,audio.input.transcription, andaudio.input.noise_reduction.-
session: RealtimeTranslationSessionUpdateRequestTranslation session fields to update. The session
typeandmodelare set at creation and cannot be changed withsession.update.-
audio: Optional[Audio]Configuration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction. Set to
nullto disable it.-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model to use for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
-
type: Literal["session.update"]The event type, must be
session.update."session.update"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Realtime Translation Session Update Request
-
class RealtimeTranslationSessionUpdateRequest: …Realtime translation session fields that can be updated with
session.update.-
audio: Optional[Audio]Configuration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction. Set to
nullto disable it.-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model to use for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
Realtime Translation Session Updated Event
-
class RealtimeTranslationSessionUpdatedEvent: …Returned when a translation session is updated with a
session.updateevent, unless there is an error.-
event_id: strThe unique ID of the server event.
-
session: RealtimeTranslationSessionThe translation session configuration.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
audio: AudioConfiguration for translation input and output audio.
-
input: Optional[AudioInput]-
noise_reduction: Optional[AudioInputNoiseReduction]Optional input noise reduction.
-
type: NoiseReductionTypeType of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioInputTranscription]Optional source-language transcription. When configured, the server emits
session.input_transcript.deltaevents. Translation itself still runs from the input audio stream.-
model: strThe transcription model used for source transcript deltas.
-
-
-
output: Optional[AudioOutput]-
language: Optional[str]Target language for translated output audio and transcript deltas.
-
-
-
expires_at: intExpiration timestamp for the session, in seconds since epoch.
-
model: strThe Realtime translation model used for this session. This field is set at session creation and cannot be changed with
session.update. -
type: Literal["translation"]The session type. Always
translationfor Realtime translation sessions."translation"
-
-
type: Literal["session.updated"]The event type, must be
session.updated."session.updated"
-
Realtime Truncation
-
RealtimeTruncationWhen the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
Realtime Truncation Retention Ratio
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
Response Audio Delta Event
-
class ResponseAudioDeltaEvent: …Returned when the model-generated audio is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strBase64-encoded audio data delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio.delta"]The event type, must be
response.output_audio.delta."response.output_audio.delta"
-
Response Audio Done Event
-
class ResponseAudioDoneEvent: …Returned when the model-generated audio is done. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio.done"]The event type, must be
response.output_audio.done."response.output_audio.done"
-
Response Audio Transcript Delta Event
-
class ResponseAudioTranscriptDeltaEvent: …Returned when the model-generated transcription of audio output is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strThe transcript delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_audio_transcript.delta"]The event type, must be
response.output_audio_transcript.delta."response.output_audio_transcript.delta"
-
Response Audio Transcript Done Event
-
class ResponseAudioTranscriptDoneEvent: …Returned when the model-generated transcription of audio output is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
transcript: strThe final transcript of the audio.
-
type: Literal["response.output_audio_transcript.done"]The event type, must be
response.output_audio_transcript.done."response.output_audio_transcript.done"
-
Response Cancel Event
-
class ResponseCancelEvent: …Send this event to cancel an in-progress response. The server will respond with a
response.doneevent with a status ofresponse.status=cancelled. If there is no response to cancel, the server will respond with an error. It's safe to callresponse.canceleven if no response is in progress, an error will be returned the session will remain unaffected.-
type: Literal["response.cancel"]The event type, must be
response.cancel."response.cancel"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
response_id: Optional[str]A specific response ID to cancel - if not provided, will cancel an in-progress response in the default conversation.
-
Response Content Part Added Event
-
class ResponseContentPartAddedEvent: …Returned when a new content part is added to an assistant message item during response generation.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item to which the content part was added.
-
output_index: intThe index of the output item in the response.
-
part: PartThe content part that was added.
-
audio: Optional[str]Base64-encoded audio data (if type is "audio").
-
text: Optional[str]The text content (if type is "text").
-
transcript: Optional[str]The transcript of the audio (if type is "audio").
-
type: Optional[Literal["text", "audio"]]The content type ("text", "audio").
-
"text" -
"audio"
-
-
-
response_id: strThe ID of the response.
-
type: Literal["response.content_part.added"]The event type, must be
response.content_part.added."response.content_part.added"
-
Response Content Part Done Event
-
class ResponseContentPartDoneEvent: …Returned when a content part is done streaming in an assistant message item. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
part: PartThe content part that is done.
-
audio: Optional[str]Base64-encoded audio data (if type is "audio").
-
text: Optional[str]The text content (if type is "text").
-
transcript: Optional[str]The transcript of the audio (if type is "audio").
-
type: Optional[Literal["text", "audio"]]The content type ("text", "audio").
-
"text" -
"audio"
-
-
-
response_id: strThe ID of the response.
-
type: Literal["response.content_part.done"]The event type, must be
response.content_part.done."response.content_part.done"
-
Response Create Event
-
class ResponseCreateEvent: …This event instructs the server to create a Response, which means triggering model inference. When in Server VAD mode, the server will create Responses automatically.
A Response will include at least one Item, and may have two, in which case the second will be a function call. These Items will be appended to the conversation history by default.
The server will respond with a
response.createdevent, events for Items and content created, and finally aresponse.doneevent to indicate the Response is complete.The
response.createevent includes inference configuration likeinstructionsandtools. If these are set, they will override the Session's configuration for this Response only.Responses can be created out-of-band of the default Conversation, meaning that they can have arbitrary input, and it's possible to disable writing the output to the Conversation. Only one Response can write to the default Conversation at a time, but otherwise multiple Responses can be created in parallel. The
metadatafield is a good way to disambiguate multiple simultaneous Responses.Clients can set
conversationtononeto create a Response that does not write to the default Conversation. Arbitrary input can be provided with theinputfield, which is an array accepting raw Items and references to existing Items.-
type: Literal["response.create"]The event type, must be
response.create."response.create"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
response: Optional[RealtimeResponseCreateParams]Create a new Realtime response with these parameters
-
audio: Optional[RealtimeResponseCreateAudioOutput]Configuration for audio input and output.
-
output: Optional[Output]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[OutputVoice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class OutputVoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
conversation: Optional[Union[str, Literal["auto", "none"], null]]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
str -
Literal["auto", "none"]Controls which conversation the response is added to. Currently supports
autoandnone, withautoas the default value. Theautovalue means that the contents of the response will be added to the default conversation. Set this tononeto create an out-of-band response which will not add items to default conversation.-
"auto" -
"none"
-
-
-
input: Optional[List[ConversationItem]]Input items to include in the prompt for the model. Using this field creates a new context for this Response instead of using the default conversation. An empty array
[]will clear the context for this Response. Note that this can include references to items that previously appeared in the session using their id.-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior. Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class RealtimeResponseCreateMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[AllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class AllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[RequireApproval]Specify which of the MCP server's tools require approval.
-
class RequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[RequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[RequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
-
Response Created Event
-
class ResponseCreatedEvent: …Returned when a new Response is created. The first event of response creation, where the response is in an initial state of
in_progress.-
event_id: strThe unique ID of the server event.
-
response: RealtimeResponseThe response resource.
-
id: Optional[str]The unique ID of the response, will look like
resp_1234. -
audio: Optional[Audio]Configuration for audio output.
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
conversation_id: Optional[str]Which conversation the response is added to, determined by the
conversationfield in theresponse.createevent. Ifauto, the response will be added to the default conversation and the value ofconversation_idwill be an id likeconv_1234. Ifnone, the response will not be added to any conversation and the value ofconversation_idwill benull. If responses are being triggered automatically by VAD the response will be added to the default conversation -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls, that was used in this response.
-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
object: Optional[Literal["realtime.response"]]The object type, must be
realtime.response."realtime.response"
-
output: Optional[List[ConversationItem]]The list of output items generated by the response.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
status: Optional[Literal["completed", "cancelled", "failed", 2 more]]The final status of the response (
completed,cancelled,failed, orincomplete,in_progress).-
"completed" -
"cancelled" -
"failed" -
"incomplete" -
"in_progress"
-
-
status_details: Optional[RealtimeResponseStatus]Additional details about the status.
-
error: Optional[Error]A description of the error that caused the response to fail, populated when the
statusisfailed.-
code: Optional[str]Error code, if any.
-
type: Optional[str]The type of error.
-
-
reason: Optional[Literal["turn_detected", "client_cancelled", "max_output_tokens", "content_filter"]]The reason the Response did not complete. For a
cancelledResponse, one ofturn_detected(the server VAD detected a new start of speech) orclient_cancelled(the client sent a cancel event). For anincompleteResponse, one ofmax_output_tokensorcontent_filter(the server-side safety filter activated and cut off the response).-
"turn_detected" -
"client_cancelled" -
"max_output_tokens" -
"content_filter"
-
-
type: Optional[Literal["completed", "cancelled", "incomplete", "failed"]]The type of error that caused the response to fail, corresponding with the
statusfield (completed,cancelled,incomplete,failed).-
"completed" -
"cancelled" -
"incomplete" -
"failed"
-
-
-
usage: Optional[RealtimeResponseUsage]Usage statistics for the Response, this will correspond to billing. A Realtime API session will maintain a conversation context and append new Items to the Conversation, thus output from previous turns (text and audio tokens) will become the input for later turns.
-
input_token_details: Optional[RealtimeResponseUsageInputTokenDetails]Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
-
input_tokens: Optional[int]The number of input tokens used in the Response, including text and audio tokens.
-
output_token_details: Optional[RealtimeResponseUsageOutputTokenDetails]Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
-
output_tokens: Optional[int]The number of output tokens sent in the Response, including text and audio tokens.
-
total_tokens: Optional[int]The total number of tokens in the Response including input and output text and audio tokens.
-
-
-
type: Literal["response.created"]The event type, must be
response.created."response.created"
-
Response Done Event
-
class ResponseDoneEvent: …Returned when a Response is done streaming. Always emitted, no matter the final state. The Response object included in the
response.doneevent will include all output Items in the Response but will omit the raw audio data.Clients should check the
statusfield of the Response to determine if it was successful (completed) or if there was another outcome:cancelled,failed, orincomplete.A response will contain all output items that were generated during the response, excluding any audio content.
-
event_id: strThe unique ID of the server event.
-
response: RealtimeResponseThe response resource.
-
id: Optional[str]The unique ID of the response, will look like
resp_1234. -
audio: Optional[Audio]Configuration for audio output.
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
conversation_id: Optional[str]Which conversation the response is added to, determined by the
conversationfield in theresponse.createevent. Ifauto, the response will be added to the default conversation and the value ofconversation_idwill be an id likeconv_1234. Ifnone, the response will not be added to any conversation and the value ofconversation_idwill benull. If responses are being triggered automatically by VAD the response will be added to the default conversation -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls, that was used in this response.
-
int -
Literal["inf"]"inf"
-
-
metadata: Optional[Metadata]Set of 16 key-value pairs that can be attached to an object. This can be useful for storing additional information about the object in a structured format, and querying for objects via API or the dashboard.
Keys are strings with a maximum length of 64 characters. Values are strings with a maximum length of 512 characters.
-
object: Optional[Literal["realtime.response"]]The object type, must be
realtime.response."realtime.response"
-
output: Optional[List[ConversationItem]]The list of output items generated by the response.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model used to respond, currently the only possible values are
[\"audio\"],[\"text\"]. Audio output always include a text transcript. Setting the output to modetextwill disable audio output from the model.-
"text" -
"audio"
-
-
status: Optional[Literal["completed", "cancelled", "failed", 2 more]]The final status of the response (
completed,cancelled,failed, orincomplete,in_progress).-
"completed" -
"cancelled" -
"failed" -
"incomplete" -
"in_progress"
-
-
status_details: Optional[RealtimeResponseStatus]Additional details about the status.
-
error: Optional[Error]A description of the error that caused the response to fail, populated when the
statusisfailed.-
code: Optional[str]Error code, if any.
-
type: Optional[str]The type of error.
-
-
reason: Optional[Literal["turn_detected", "client_cancelled", "max_output_tokens", "content_filter"]]The reason the Response did not complete. For a
cancelledResponse, one ofturn_detected(the server VAD detected a new start of speech) orclient_cancelled(the client sent a cancel event). For anincompleteResponse, one ofmax_output_tokensorcontent_filter(the server-side safety filter activated and cut off the response).-
"turn_detected" -
"client_cancelled" -
"max_output_tokens" -
"content_filter"
-
-
type: Optional[Literal["completed", "cancelled", "incomplete", "failed"]]The type of error that caused the response to fail, corresponding with the
statusfield (completed,cancelled,incomplete,failed).-
"completed" -
"cancelled" -
"incomplete" -
"failed"
-
-
-
usage: Optional[RealtimeResponseUsage]Usage statistics for the Response, this will correspond to billing. A Realtime API session will maintain a conversation context and append new Items to the Conversation, thus output from previous turns (text and audio tokens) will become the input for later turns.
-
input_token_details: Optional[RealtimeResponseUsageInputTokenDetails]Details about the input tokens used in the Response. Cached tokens are tokens from previous turns in the conversation that are included as context for the current response. Cached tokens here are counted as a subset of input tokens, meaning input tokens will include cached and uncached tokens.
-
audio_tokens: Optional[int]The number of audio tokens used as input for the Response.
-
cached_tokens: Optional[int]The number of cached tokens used as input for the Response.
-
cached_tokens_details: Optional[CachedTokensDetails]Details about the cached tokens used as input for the Response.
-
audio_tokens: Optional[int]The number of cached audio tokens used as input for the Response.
-
image_tokens: Optional[int]The number of cached image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of cached text tokens used as input for the Response.
-
-
image_tokens: Optional[int]The number of image tokens used as input for the Response.
-
text_tokens: Optional[int]The number of text tokens used as input for the Response.
-
-
input_tokens: Optional[int]The number of input tokens used in the Response, including text and audio tokens.
-
output_token_details: Optional[RealtimeResponseUsageOutputTokenDetails]Details about the output tokens used in the Response.
-
audio_tokens: Optional[int]The number of audio tokens used in the Response.
-
text_tokens: Optional[int]The number of text tokens used in the Response.
-
-
output_tokens: Optional[int]The number of output tokens sent in the Response, including text and audio tokens.
-
total_tokens: Optional[int]The total number of tokens in the Response including input and output text and audio tokens.
-
-
-
type: Literal["response.done"]The event type, must be
response.done."response.done"
-
Response Function Call Arguments Delta Event
-
class ResponseFunctionCallArgumentsDeltaEvent: …Returned when the model-generated function call arguments are updated.
-
call_id: strThe ID of the function call.
-
delta: strThe arguments delta as a JSON string.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the function call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.function_call_arguments.delta"]The event type, must be
response.function_call_arguments.delta."response.function_call_arguments.delta"
-
Response Function Call Arguments Done Event
-
class ResponseFunctionCallArgumentsDoneEvent: …Returned when the model-generated function call arguments are done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
arguments: strThe final arguments as a JSON string.
-
call_id: strThe ID of the function call.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the function call item.
-
name: strThe name of the function that was called.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.function_call_arguments.done"]The event type, must be
response.function_call_arguments.done."response.function_call_arguments.done"
-
Response Mcp Call Arguments Delta
-
class ResponseMcpCallArgumentsDelta: …Returned when MCP tool call arguments are updated during response generation.
-
delta: strThe JSON-encoded arguments delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.mcp_call_arguments.delta"]The event type, must be
response.mcp_call_arguments.delta."response.mcp_call_arguments.delta"
-
obfuscation: Optional[str]If present, indicates the delta text was obfuscated.
-
Response Mcp Call Arguments Done
-
class ResponseMcpCallArgumentsDone: …Returned when MCP tool call arguments are finalized during response generation.
-
arguments: strThe final JSON-encoded arguments string.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.mcp_call_arguments.done"]The event type, must be
response.mcp_call_arguments.done."response.mcp_call_arguments.done"
-
Response Mcp Call Completed
-
class ResponseMcpCallCompleted: …Returned when an MCP tool call has completed successfully.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.completed"]The event type, must be
response.mcp_call.completed."response.mcp_call.completed"
-
Response Mcp Call Failed
-
class ResponseMcpCallFailed: …Returned when an MCP tool call has failed.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.failed"]The event type, must be
response.mcp_call.failed."response.mcp_call.failed"
-
Response Mcp Call In Progress
-
class ResponseMcpCallInProgress: …Returned when an MCP tool call has started and is in progress.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the MCP tool call item.
-
output_index: intThe index of the output item in the response.
-
type: Literal["response.mcp_call.in_progress"]The event type, must be
response.mcp_call.in_progress."response.mcp_call.in_progress"
-
Response Output Item Added Event
-
class ResponseOutputItemAddedEvent: …Returned when a new Item is created during Response generation.
-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
output_index: intThe index of the output item in the Response.
-
response_id: strThe ID of the Response to which the item belongs.
-
type: Literal["response.output_item.added"]The event type, must be
response.output_item.added."response.output_item.added"
-
Response Output Item Done Event
-
class ResponseOutputItemDoneEvent: …Returned when an Item is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
event_id: strThe unique ID of the server event.
-
item: ConversationItemA single item within a Realtime conversation.
-
class RealtimeConversationItemSystemMessage: …A system message in a Realtime conversation can be used to provide additional context or instructions to the model. This is similar but distinct from the instruction prompt provided at the start of a conversation, as system messages can be added at any point in the conversation. For major changes to the conversation's behavior, use instructions, but for smaller updates (e.g. "the user is now asking about a different topic"), use system messages.
-
content: List[Content]The content of the message.
-
text: Optional[str]The text content.
-
type: Optional[Literal["input_text"]]The content type. Always
input_textfor system messages."input_text"
-
-
role: Literal["system"]The role of the message sender. Always
system."system"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemUserMessage: …A user message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes (for
input_audio), these will be parsed as the format specified in the session input audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified. -
detail: Optional[Literal["auto", "low", "high"]]The detail level of the image (for
input_image).autowill default tohigh.-
"auto" -
"low" -
"high"
-
-
image_url: Optional[str]Base64-encoded image bytes (for
input_image) as a data URI. For exampledata:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA.... Supported formats are PNG and JPEG. -
text: Optional[str]The text content (for
input_text). -
transcript: Optional[str]Transcript of the audio (for
input_audio). This is not sent to the model, but will be attached to the message item for reference. -
type: Optional[Literal["input_text", "input_audio", "input_image"]]The content type (
input_text,input_audio, orinput_image).-
"input_text" -
"input_audio" -
"input_image"
-
-
-
role: Literal["user"]The role of the message sender. Always
user."user"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemAssistantMessage: …An assistant message item in a Realtime conversation.
-
content: List[Content]The content of the message.
-
audio: Optional[str]Base64-encoded audio bytes, these will be parsed as the format specified in the session output audio type configuration. This defaults to PCM 16-bit 24kHz mono if not specified.
-
text: Optional[str]The text content.
-
transcript: Optional[str]The transcript of the audio content, this will always be present if the output type is
audio. -
type: Optional[Literal["output_text", "output_audio"]]The content type,
output_textoroutput_audiodepending on the sessionoutput_modalitiesconfiguration.-
"output_text" -
"output_audio"
-
-
-
role: Literal["assistant"]The role of the message sender. Always
assistant."assistant"
-
type: Literal["message"]The type of the item. Always
message."message"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCall: …A function call item in a Realtime conversation.
-
arguments: strThe arguments of the function call. This is a JSON-encoded string representing the arguments passed to the function, for example
{"arg1": "value1", "arg2": 42}. -
name: strThe name of the function being called.
-
type: Literal["function_call"]The type of the item. Always
function_call."function_call"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
call_id: Optional[str]The ID of the function call.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeConversationItemFunctionCallOutput: …A function call output item in a Realtime conversation.
-
call_id: strThe ID of the function call this output is for.
-
output: strThe output of the function call, this is free text and can contain any information or simply be empty.
-
type: Literal["function_call_output"]The type of the item. Always
function_call_output."function_call_output"
-
id: Optional[str]The unique ID of the item. This may be provided by the client or generated by the server.
-
object: Optional[Literal["realtime.item"]]Identifier for the API object being returned - always
realtime.item. Optional when creating a new item."realtime.item"
-
status: Optional[Literal["completed", "incomplete", "in_progress"]]The status of the item. Has no effect on the conversation.
-
"completed" -
"incomplete" -
"in_progress"
-
-
-
class RealtimeMcpApprovalResponse: …A Realtime item responding to an MCP approval request.
-
id: strThe unique ID of the approval response.
-
approval_request_id: strThe ID of the approval request being answered.
-
approve: boolWhether the request was approved.
-
type: Literal["mcp_approval_response"]The type of the item. Always
mcp_approval_response."mcp_approval_response"
-
reason: Optional[str]Optional reason for the decision.
-
-
class RealtimeMcpListTools: …A Realtime item listing tools available on an MCP server.
-
server_label: strThe label of the MCP server.
-
tools: List[Tool]The tools available on the server.
-
input_schema: objectThe JSON schema describing the tool's input.
-
name: strThe name of the tool.
-
annotations: Optional[object]Additional annotations about the tool.
-
description: Optional[str]The description of the tool.
-
-
type: Literal["mcp_list_tools"]The type of the item. Always
mcp_list_tools."mcp_list_tools"
-
id: Optional[str]The unique ID of the list.
-
-
class RealtimeMcpToolCall: …A Realtime item representing an invocation of a tool on an MCP server.
-
id: strThe unique ID of the tool call.
-
arguments: strA JSON string of the arguments passed to the tool.
-
name: strThe name of the tool that was run.
-
server_label: strThe label of the MCP server running the tool.
-
type: Literal["mcp_call"]The type of the item. Always
mcp_call."mcp_call"
-
approval_request_id: Optional[str]The ID of an associated approval request, if any.
-
error: Optional[Error]The error from the tool call, if any.
-
class RealtimeMcpProtocolError: …-
code: int -
message: str -
type: Literal["protocol_error"]"protocol_error"
-
-
class RealtimeMcpToolExecutionError: …-
message: str -
type: Literal["tool_execution_error"]"tool_execution_error"
-
-
class RealtimeMcphttpError: …-
code: int -
message: str -
type: Literal["http_error"]"http_error"
-
-
-
output: Optional[str]The output from the tool call.
-
-
class RealtimeMcpApprovalRequest: …A Realtime item requesting human approval of a tool invocation.
-
id: strThe unique ID of the approval request.
-
arguments: strA JSON string of arguments for the tool.
-
name: strThe name of the tool to run.
-
server_label: strThe label of the MCP server making the request.
-
type: Literal["mcp_approval_request"]The type of the item. Always
mcp_approval_request."mcp_approval_request"
-
-
-
output_index: intThe index of the output item in the Response.
-
response_id: strThe ID of the Response to which the item belongs.
-
type: Literal["response.output_item.done"]The event type, must be
response.output_item.done."response.output_item.done"
-
Response Text Delta Event
-
class ResponseTextDeltaEvent: …Returned when the text value of an "output_text" content part is updated.
-
content_index: intThe index of the content part in the item's content array.
-
delta: strThe text delta.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
type: Literal["response.output_text.delta"]The event type, must be
response.output_text.delta."response.output_text.delta"
-
Response Text Done Event
-
class ResponseTextDoneEvent: …Returned when the text value of an "output_text" content part is done streaming. Also emitted when a Response is interrupted, incomplete, or cancelled.
-
content_index: intThe index of the content part in the item's content array.
-
event_id: strThe unique ID of the server event.
-
item_id: strThe ID of the item.
-
output_index: intThe index of the output item in the response.
-
response_id: strThe ID of the response.
-
text: strThe final text content.
-
type: Literal["response.output_text.done"]The event type, must be
response.output_text.done."response.output_text.done"
-
Session Created Event
-
class SessionCreatedEvent: …Returned when a Session is created. Emitted automatically when a new connection is established as the first server event. This event will contain the default Session configuration.
-
event_id: strThe unique ID of the server event.
-
session: SessionThe session configuration.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
-
type: Literal["session.created"]The event type, must be
session.created."session.created"
-
Session Update Event
-
class SessionUpdateEvent: …Send this event to update the session’s configuration. The client may send this event at any time to update any field except for
voiceandmodel.voicecan be updated only if there have been no other audio outputs yet.When the server receives a
session.update, it will respond with asession.updatedevent showing the full, effective configuration. Only the fields that are present in thesession.updateare updated. To clear a field likeinstructions, pass an empty string. To clear a field liketools, pass an empty array. To clear a field liketurn_detection, passnull.-
session: SessionUpdate the Realtime session. Choose either a realtime session or a transcription session.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
-
type: Literal["session.update"]The event type, must be
session.update."session.update"
-
event_id: Optional[str]Optional client-generated ID used to identify this event. This is an arbitrary string that a client may assign. It will be passed back if there is an error with the event, but the corresponding
session.updatedevent will not include it.
-
Session Updated Event
-
class SessionUpdatedEvent: …Returned when a session is updated with a
session.updateevent, unless there is an error.-
event_id: strThe unique ID of the server event.
-
session: SessionThe session configuration.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
-
type: Literal["session.updated"]The event type, must be
session.updated."session.updated"
-
Transcription Session Update
-
class TranscriptionSessionUpdate: …Send this event to update a transcription session.
-
session: SessionRealtime transcription session object configuration.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]The set of items to include in the transcription. Current available items are:
item.input_audio_transcription.logprobs"item.input_audio_transcription.logprobs"
-
input_audio_format: Optional[Literal["pcm16", "g711_ulaw", "g711_alaw"]]The format of input audio. Options are
pcm16,g711_ulaw, org711_alaw. Forpcm16, input audio must be 16-bit PCM at a 24kHz sample rate, single channel (mono), and little-endian byte order.-
"pcm16" -
"g711_ulaw" -
"g711_alaw"
-
-
input_audio_noise_reduction: Optional[SessionInputAudioNoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
input_audio_transcription: Optional[AudioTranscription]Configuration for input audio transcription. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.
-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[SessionTurnDetection]Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[Literal["server_vad"]]Type of turn detection. Only
server_vadis currently supported for transcription sessions."server_vad"
-
-
-
type: Literal["transcription_session.update"]The event type, must be
transcription_session.update."transcription_session.update"
-
event_id: Optional[str]Optional client-generated ID used to identify this event.
-
Transcription Session Updated Event
-
class TranscriptionSessionUpdatedEvent: …Returned when a transcription session is updated with a
transcription_session.updateevent, unless there is an error.-
event_id: strThe unique ID of the server event.
-
session: SessionA new Realtime transcription session configuration.
When a session is created on the server via REST API, the session object also contains an ephemeral key. Default TTL for keys is 10 minutes. This property is not present when a session is updated via the WebSocket API.
-
client_secret: SessionClientSecretEphemeral key returned by the API. Only present when the session is created on the server via REST API.
-
expires_at: intTimestamp for when the token expires. Currently, all tokens expire after one minute.
-
value: strEphemeral key usable in client environments to authenticate connections to the Realtime API. Use this in client-side environments rather than a standard API token, which should only be used server-side.
-
-
input_audio_format: Optional[str]The format of input audio. Options are
pcm16,g711_ulaw, org711_alaw. -
input_audio_transcription: Optional[AudioTranscription]-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. To disable audio, set this to ["text"].
-
"text" -
"audio"
-
-
turn_detection: Optional[SessionTurnDetection]Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[str]Type of turn detection, only
server_vadis currently supported.
-
-
-
type: Literal["transcription_session.updated"]The event type, must be
transcription_session.updated."transcription_session.updated"
-
Client Secrets
Create client secret
realtime.client_secrets.create(ClientSecretCreateParams**kwargs) -> ClientSecretCreateResponse
post /realtime/client_secrets
Create a Realtime client secret with an associated session configuration.
Client secrets are short-lived tokens that can be passed to a client app, such as a web frontend or mobile client, which grants access to the Realtime API without leaking your main API key. You can configure a custom TTL for each client secret.
You can also attach session configuration options to the client secret, which will be applied to any sessions created using that client secret, but these can also be overridden by the client connection.
Learn more about authentication with client secrets over WebRTC.
Returns the created client secret and the effective session object. The client secret is a string that looks like ek_1234.
Parameters
-
expires_after: Optional[ExpiresAfter]Configuration for the client secret expiration. Expiration refers to the time after which a client secret will no longer be valid for creating sessions. The session itself may continue after that time once started. A secret can be used to create multiple sessions until it expires.
-
anchor: Optional[Literal["created_at"]]The anchor point for the client secret expiration, meaning that
secondswill be added to thecreated_attime of the client secret to produce an expiration timestamp. Onlycreated_atis currently supported."created_at"
-
seconds: Optional[int]The number of seconds from the anchor point to the expiration. Select a value between
10and7200(2 hours). This default to 600 seconds (10 minutes) if not specified.
-
-
session: Optional[Session]Session configuration to use for the client secret. Choose either a realtime session or a transcription session.
-
class RealtimeSessionCreateRequest: …Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateRequest: …Realtime transcription session object configuration.
-
type: Literal["transcription"]The type of session to create. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[RealtimeTranscriptionSessionAudio]Configuration for input and output audio.
-
input: Optional[RealtimeTranscriptionSessionAudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. -
turn_detection: Optional[RealtimeTranscriptionSessionAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
-
Returns
-
class ClientSecretCreateResponse: …Response from creating a session and client secret for the Realtime API.
-
expires_at: intExpiration timestamp for the client secret, in seconds since epoch.
-
session: SessionThe session configuration for either a realtime or transcription session.
-
class RealtimeSessionCreateResponse: …A Realtime session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: Literal["realtime.session"]The object type. Always
realtime.session."realtime.session"
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[Audio]Configuration for input and output audio.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[AudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class AudioInputTurnDetectionServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class AudioInputTurnDetectionSemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class ToolMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[ToolMcpToolAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class ToolMcpToolAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[ToolMcpToolRequireApproval]Specify which of the MCP server's tools require approval.
-
class ToolMcpToolRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[Tracing]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingTracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateResponse: …A Realtime transcription session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: strThe object type. Always
realtime.transcription_session. -
type: Literal["transcription"]The type of session. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[Audio]Configuration for input audio for the session.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction.
-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription] -
turn_detection: Optional[RealtimeTranscriptionSessionTurnDetection]Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech. Forgpt-realtime-whisper, this must benull; VAD is not supported.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[str]Type of turn detection, only
server_vadis currently supported.
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
-
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription. -
"item.input_audio_transcription.logprobs"
-
-
-
-
value: strThe generated client secret value.
-
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client_secret = client.realtime.client_secrets.create()
print(client_secret.expires_at)
Response
{
"expires_at": 0,
"session": {
"id": "id",
"object": "realtime.session",
"type": "realtime",
"audio": {
"input": {
"format": {
"rate": 24000,
"type": "audio/pcm"
},
"noise_reduction": {
"type": "near_field"
},
"transcription": {
"delay": "minimal",
"language": "language",
"model": "whisper-1",
"prompt": "prompt"
},
"turn_detection": {
"type": "server_vad",
"create_response": true,
"idle_timeout_ms": 5000,
"interrupt_response": true,
"prefix_padding_ms": 0,
"silence_duration_ms": 0,
"threshold": 0
}
},
"output": {
"format": {
"rate": 24000,
"type": "audio/pcm"
},
"speed": 0.25,
"voice": "ash"
}
},
"expires_at": 0,
"include": [
"item.input_audio_transcription.logprobs"
],
"instructions": "instructions",
"max_output_tokens": "inf",
"model": "gpt-realtime",
"output_modalities": [
"text"
],
"prompt": {
"id": "id",
"variables": {
"foo": "string"
},
"version": "version"
},
"reasoning": {
"effort": "minimal"
},
"tool_choice": "none",
"tools": [
{
"description": "description",
"name": "name",
"parameters": {},
"type": "function"
}
],
"tracing": "auto",
"truncation": "auto"
},
"value": "value"
}
Domain Types
Realtime Session Create Response
-
class RealtimeSessionCreateResponse: …A Realtime session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: Literal["realtime.session"]The object type. Always
realtime.session."realtime.session"
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[Audio]Configuration for input and output audio.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[AudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class AudioInputTurnDetectionServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class AudioInputTurnDetectionSemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class ToolMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[ToolMcpToolAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class ToolMcpToolAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[ToolMcpToolRequireApproval]Specify which of the MCP server's tools require approval.
-
class ToolMcpToolRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[Tracing]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingTracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
Realtime Transcription Session Create Response
-
class RealtimeTranscriptionSessionCreateResponse: …A Realtime transcription session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: strThe object type. Always
realtime.transcription_session. -
type: Literal["transcription"]The type of session. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[Audio]Configuration for input audio for the session.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction.
-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeTranscriptionSessionTurnDetection]Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech. Forgpt-realtime-whisper, this must benull; VAD is not supported.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[str]Type of turn detection, only
server_vadis currently supported.
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
-
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription. -
"item.input_audio_transcription.logprobs"
-
-
Realtime Transcription Session Turn Detection
-
class RealtimeTranscriptionSessionTurnDetection: …Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech. Forgpt-realtime-whisper, this must benull; VAD is not supported.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[str]Type of turn detection, only
server_vadis currently supported.
-
Client Secret Create Response
-
class ClientSecretCreateResponse: …Response from creating a session and client secret for the Realtime API.
-
expires_at: intExpiration timestamp for the client secret, in seconds since epoch.
-
session: SessionThe session configuration for either a realtime or transcription session.
-
class RealtimeSessionCreateResponse: …A Realtime session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: Literal["realtime.session"]The object type. Always
realtime.session."realtime.session"
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[Audio]Configuration for input and output audio.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[AudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class AudioInputTurnDetectionServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class AudioInputTurnDetectionSemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[AudioOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Union[str, Literal["alloy", "ash", "ballad", 7 more], null]]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]The voice the model uses to respond. Voice cannot be changed during the session once the model has responded with audio at least once. Current voice options are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. We recommendmarinandcedarfor best quality.-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[ToolChoice]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[List[Tool]]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class ToolMcpTool: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[ToolMcpToolAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class ToolMcpToolAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[ToolMcpToolRequireApproval]Specify which of the MCP server's tools require approval.
-
class ToolMcpToolRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[ToolMcpToolRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[Tracing]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingTracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
-
class RealtimeTranscriptionSessionCreateResponse: …A Realtime transcription session configuration object.
-
id: strUnique identifier for the session that looks like
sess_1234567890abcdef. -
object: strThe object type. Always
realtime.transcription_session. -
type: Literal["transcription"]The type of session. Always
transcriptionfor transcription sessions."transcription"
-
audio: Optional[Audio]Configuration for input audio for the session.
-
input: Optional[AudioInput]-
format: Optional[RealtimeAudioFormats]The PCM audio format. Only a 24kHz sample rate is supported.
-
noise_reduction: Optional[AudioInputNoiseReduction]Configuration for input audio noise reduction.
-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.
-
-
transcription: Optional[AudioTranscription] -
turn_detection: Optional[RealtimeTranscriptionSessionTurnDetection]Configuration for turn detection. Can be set to
nullto turn off. Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech. Forgpt-realtime-whisper, this must benull; VAD is not supported.-
prefix_padding_ms: Optional[int]Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.
-
silence_duration_ms: Optional[int]Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user.
-
threshold: Optional[float]Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
type: Optional[str]Type of turn detection, only
server_vadis currently supported.
-
-
-
-
expires_at: Optional[int]Expiration timestamp for the session, in seconds since epoch.
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
-
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription. -
"item.input_audio_transcription.logprobs"
-
-
-
-
value: strThe generated client secret value.
-
Calls
Accept call
realtime.calls.accept(strcall_id, CallAcceptParams**kwargs)
post /realtime/calls/{call_id}/accept
Accept an incoming SIP call and configure the realtime session that will handle it.
Parameters
-
call_id: str -
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfigParam]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"]]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePromptParam]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoningParam]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfigParam]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfigParam]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfigParam]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncationParam]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client.realtime.calls.accept(
call_id="call_id",
type="realtime",
)
Hang up call
realtime.calls.hangup(strcall_id)
post /realtime/calls/{call_id}/hangup
End an active Realtime API call, whether it was initiated over SIP or WebRTC.
Parameters
call_id: str
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client.realtime.calls.hangup(
"call_id",
)
Refer call
realtime.calls.refer(strcall_id, CallReferParams**kwargs)
post /realtime/calls/{call_id}/refer
Transfer an active SIP call to a new destination using the SIP REFER verb.
Parameters
-
call_id: str -
target_uri: strURI that should appear in the SIP Refer-To header. Supports values like
tel:+14155550123orsip:agent@example.com.
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client.realtime.calls.refer(
call_id="call_id",
target_uri="tel:+14155550123",
)
Reject call
realtime.calls.reject(strcall_id, CallRejectParams**kwargs)
post /realtime/calls/{call_id}/reject
Decline an incoming SIP call by returning a SIP status code to the caller.
Parameters
-
call_id: str -
status_code: Optional[int]SIP response code to send back to the caller. Defaults to
603(Decline) when omitted.
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
client.realtime.calls.reject(
call_id="call_id",
)
Create call
realtime.calls.create(CallCreateParams**kwargs) -> BinaryResponseContent
post /realtime/calls
Create a new Realtime API call over WebRTC and receive the SDP answer needed to complete the peer connection.
Parameters
-
sdp: strWebRTC Session Description Protocol (SDP) offer generated by the caller.
-
session: Optional[RealtimeSessionCreateRequestParam]Realtime session object configuration.
-
type: Literal["realtime"]The type of session to create. Always
realtimefor the Realtime API."realtime"
-
audio: Optional[RealtimeAudioConfig]Configuration for input and output audio.
-
input: Optional[RealtimeAudioConfigInput]-
format: Optional[RealtimeAudioFormats]The format of the input audio.
-
class AudioPCM: …The PCM audio format. Only a 24kHz sample rate is supported.
-
rate: Optional[Literal[24000]]The sample rate of the audio. Always
24000.24000
-
type: Optional[Literal["audio/pcm"]]The audio format. Always
audio/pcm."audio/pcm"
-
-
class AudioPCMU: …The G.711 μ-law format.
-
type: Optional[Literal["audio/pcmu"]]The audio format. Always
audio/pcmu."audio/pcmu"
-
-
class AudioPCMA: …The G.711 A-law format.
-
type: Optional[Literal["audio/pcma"]]The audio format. Always
audio/pcma."audio/pcma"
-
-
-
noise_reduction: Optional[NoiseReduction]Configuration for input audio noise reduction. This can be set to
nullto turn off. Noise reduction filters audio added to the input audio buffer before it is sent to VAD and the model. Filtering the audio can improve VAD and turn detection accuracy (reducing false positives) and model performance by improving perception of the input audio.-
type: Optional[NoiseReductionType]Type of noise reduction.
near_fieldis for close-talking microphones such as headphones,far_fieldis for far-field microphones such as laptop or conference room microphones.-
"near_field" -
"far_field"
-
-
-
transcription: Optional[AudioTranscription]Configuration for input audio transcription, defaults to off and can be set to
nullto turn off once on. Input audio transcription is not native to the model, since the model consumes audio directly. Transcription runs asynchronously through the /audio/transcriptions endpoint and should be treated as guidance of input audio content rather than precisely what the model heard. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.-
delay: Optional[Literal["minimal", "low", "medium", 2 more]]Controls how long the model waits before emitting transcription text. Higher values can improve transcription accuracy at the cost of latency. Only supported with
gpt-realtime-whisperin GA Realtime sessions.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
language: Optional[str]The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency. -
model: Optional[Union[str, Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more], null]]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
str -
Literal["whisper-1", "gpt-4o-mini-transcribe", "gpt-4o-mini-transcribe-2025-12-15", 3 more]The model to use for transcription. Current options are
whisper-1,gpt-4o-mini-transcribe,gpt-4o-mini-transcribe-2025-12-15,gpt-4o-transcribe,gpt-4o-transcribe-diarize, andgpt-realtime-whisper. Usegpt-4o-transcribe-diarizewhen you need diarization with speaker labels.-
"whisper-1" -
"gpt-4o-mini-transcribe" -
"gpt-4o-mini-transcribe-2025-12-15" -
"gpt-4o-transcribe" -
"gpt-4o-transcribe-diarize" -
"gpt-realtime-whisper"
-
-
-
prompt: Optional[str]An optional text to guide the model's style or continue a previous audio segment. For
whisper-1, the prompt is a list of keywords. Forgpt-4o-transcribemodels (excludinggpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported withgpt-realtime-whisperin GA Realtime sessions.
-
-
turn_detection: Optional[RealtimeAudioInputTurnDetection]Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to
nullto turn off, in which case the client must manually trigger model response.Server VAD means that the model will detect the start and end of speech based on audio volume and respond at the end of user speech.
Semantic VAD is more advanced and uses a turn detection model (in conjunction with VAD) to semantically estimate whether the user has finished speaking, then dynamically sets a timeout based on this probability. For example, if user audio trails off with "uhhm", the model will score a low probability of turn end and wait longer for the user to continue speaking. This can be useful for more natural conversations, but may have a higher latency.
For
gpt-realtime-whispertranscription sessions, turn detection must be set tonull; VAD is not supported.-
class ServerVad: …Server-side voice activity detection (VAD) which flips on when user speech is detected and off after a period of silence.
-
type: Literal["server_vad"]Type of turn detection,
server_vadto turn on simple Server VAD."server_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs. If
interrupt_responseis set tofalsethis may fail to create a response if the model is already responding.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
idle_timeout_ms: Optional[int]Optional timeout after which a model response will be triggered automatically. This is useful for situations in which a long pause from the user is unexpected, such as a phone call. The model will effectively prompt the user to continue the conversation based on the current context.
The timeout value will be applied after the last model response's audio has finished playing, i.e. it's set to the
response.donetime plus audio playback duration.An
input_audio_buffer.timeout_triggeredevent (plus events associated with the Response) will be emitted when the timeout is reached. Idle timeout is currently only supported forserver_vadmode. -
interrupt_response: Optional[bool]Whether or not to automatically interrupt (cancel) any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Iftruethen the response will be cancelled, otherwise it will continue until complete.If both
create_responseandinterrupt_responseare set tofalse, the model will never respond automatically but VAD events will still be emitted. -
prefix_padding_ms: Optional[int]Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. -
silence_duration_ms: Optional[int]Used only for
server_vadmode. Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms. With shorter values the model will respond more quickly, but may jump in on short pauses from the user. -
threshold: Optional[float]Used only for
server_vadmode. Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher threshold will require louder audio to activate the model, and thus might perform better in noisy environments.
-
-
class SemanticVad: …Server-side semantic turn detection which uses a model to determine when the user has finished speaking.
-
type: Literal["semantic_vad"]Type of turn detection,
semantic_vadto turn on Semantic VAD."semantic_vad"
-
create_response: Optional[bool]Whether or not to automatically generate a response when a VAD stop event occurs.
-
eagerness: Optional[Literal["low", "medium", "high", "auto"]]Used only for
semantic_vadmode. The eagerness of the model to respond.lowwill wait longer for the user to continue speaking,highwill respond more quickly.autois the default and is equivalent tomedium.low,medium, andhighhave max timeouts of 8s, 4s, and 2s respectively.-
"low" -
"medium" -
"high" -
"auto"
-
-
interrupt_response: Optional[bool]Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs.
-
-
-
-
output: Optional[RealtimeAudioConfigOutput]-
format: Optional[RealtimeAudioFormats]The format of the output audio.
-
speed: Optional[float]The speed of the model's spoken response as a multiple of the original speed. 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed. This value can only be changed in between model turns, not while a response is in progress.
This parameter is a post-processing adjustment to the audio after it is generated, it's also possible to prompt the model to speak faster or slower.
-
voice: Optional[Voice]The voice the model uses to respond. Supported built-in voices are
alloy,ash,ballad,coral,echo,sage,shimmer,verse,marin, andcedar. You may also provide a custom voice object with anid, for example{ "id": "voice_1234" }. Voice cannot be changed during the session once the model has responded with audio at least once. We recommendmarinandcedarfor best quality.-
str -
Literal["alloy", "ash", "ballad", 7 more]-
"alloy" -
"ash" -
"ballad" -
"coral" -
"echo" -
"sage" -
"shimmer" -
"verse" -
"marin" -
"cedar"
-
-
class VoiceID: …Custom voice reference.
-
id: strThe custom voice ID, e.g.
voice_1234.
-
-
-
-
-
include: Optional[List[Literal["item.input_audio_transcription.logprobs"]]]Additional fields to include in server outputs.
item.input_audio_transcription.logprobs: Include logprobs for input audio transcription."item.input_audio_transcription.logprobs"
-
instructions: Optional[str]The default system instructions (i.e. system message) prepended to model calls. This field allows the client to guide the model on desired responses. The model can be instructed on response content and format, (e.g. "be extremely succinct", "act friendly", "here are examples of good responses") and on audio behavior (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The instructions are not guaranteed to be followed by the model, but they provide guidance to the model on the desired behavior.
Note that the server sets default instructions which will be used if this field is not set and are visible in the
session.createdevent at the start of the session. -
max_output_tokens: Optional[Union[int, Literal["inf"], null]]Maximum number of output tokens for a single assistant response, inclusive of tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
inffor the maximum available tokens for a given model. Defaults toinf.-
int -
Literal["inf"]"inf"
-
-
model: Optional[Union[str, Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more], null]]The Realtime model used for this session.
-
str -
Literal["gpt-realtime", "gpt-realtime-1.5", "gpt-realtime-2", 14 more]The Realtime model used for this session.
-
"gpt-realtime" -
"gpt-realtime-1.5" -
"gpt-realtime-2" -
"gpt-realtime-2025-08-28" -
"gpt-4o-realtime-preview" -
"gpt-4o-realtime-preview-2024-10-01" -
"gpt-4o-realtime-preview-2024-12-17" -
"gpt-4o-realtime-preview-2025-06-03" -
"gpt-4o-mini-realtime-preview" -
"gpt-4o-mini-realtime-preview-2024-12-17" -
"gpt-realtime-mini" -
"gpt-realtime-mini-2025-10-06" -
"gpt-realtime-mini-2025-12-15" -
"gpt-audio-1.5" -
"gpt-audio-mini" -
"gpt-audio-mini-2025-10-06" -
"gpt-audio-mini-2025-12-15"
-
-
-
output_modalities: Optional[List[Literal["text", "audio"]]]The set of modalities the model can respond with. It defaults to
["audio"], indicating that the model will respond with audio plus a transcript.["text"]can be used to make the model respond with text only. It is not possible to request bothtextandaudioat the same time.-
"text" -
"audio"
-
-
parallel_tool_calls: Optional[bool]Whether the model may call multiple tools in parallel. Only supported by reasoning Realtime models such as
gpt-realtime-2. -
prompt: Optional[ResponsePrompt]Reference to a prompt template and its variables. Learn more.
-
id: strThe unique identifier of the prompt template to use.
-
variables: Optional[Dict[str, Variables]]Optional map of values to substitute in for variables in your prompt. The substitution values can either be strings, or other Response input types like images or files.
-
str -
class ResponseInputText: …A text input to the model.
-
text: strThe text input to the model.
-
type: Literal["input_text"]The type of the input item. Always
input_text."input_text"
-
-
class ResponseInputImage: …An image input to the model. Learn about image inputs.
-
detail: Literal["low", "high", "auto", "original"]The detail level of the image to be sent to the model. One of
high,low,auto, ororiginal. Defaults toauto.-
"low" -
"high" -
"auto" -
"original"
-
-
type: Literal["input_image"]The type of the input item. Always
input_image."input_image"
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
image_url: Optional[str]The URL of the image to be sent to the model. A fully qualified URL or base64 encoded image in a data URL.
-
-
class ResponseInputFile: …A file input to the model.
-
type: Literal["input_file"]The type of the input item. Always
input_file."input_file"
-
detail: Optional[Literal["low", "high"]]The detail level of the file to be sent to the model. Use
lowfor the default rendering behavior, orhighto render the file at higher quality. Defaults tolow.-
"low" -
"high"
-
-
file_data: Optional[str]The content of the file to be sent to the model.
-
file_id: Optional[str]The ID of the file to be sent to the model.
-
file_url: Optional[str]The URL of the file to be sent to the model.
-
filename: Optional[str]The name of the file to be sent to the model.
-
-
-
version: Optional[str]Optional version of the prompt template.
-
-
reasoning: Optional[RealtimeReasoning]Configuration for reasoning-capable Realtime models such as
gpt-realtime-2.-
effort: Optional[RealtimeReasoningEffort]Constrains effort on reasoning for reasoning-capable Realtime models such as
gpt-realtime-2.-
"minimal" -
"low" -
"medium" -
"high" -
"xhigh"
-
-
-
tool_choice: Optional[RealtimeToolChoiceConfig]How the model chooses tools. Provide one of the string modes or force a specific function/MCP tool.
-
Literal["none", "auto", "required"]-
"none" -
"auto" -
"required"
-
-
class ToolChoiceFunction: …Use this option to force the model to call a specific function.
-
name: strThe name of the function to call.
-
type: Literal["function"]For function calling, the type is always
function."function"
-
-
class ToolChoiceMcp: …Use this option to force the model to call a specific tool on a remote MCP server.
-
server_label: strThe label of the MCP server to use.
-
type: Literal["mcp"]For MCP tools, the type is always
mcp."mcp"
-
name: Optional[str]The name of the tool to call on the server.
-
-
-
tools: Optional[RealtimeToolsConfig]Tools available to the model.
-
class RealtimeFunctionTool: …-
description: Optional[str]The description of the function, including guidance on when and how to call it, and guidance about what to tell the user when calling (if anything).
-
name: Optional[str]The name of the function.
-
parameters: Optional[object]Parameters of the function in JSON Schema.
-
type: Optional[Literal["function"]]The type of the tool, i.e.
function."function"
-
-
class Mcp: …Give the model access to additional tools via remote Model Context Protocol (MCP) servers. Learn more about MCP.
-
server_label: strA label for this MCP server, used to identify it in tool calls.
-
type: Literal["mcp"]The type of the MCP tool. Always
mcp."mcp"
-
allowed_tools: Optional[McpAllowedTools]List of allowed tool names or a filter object.
-
List[str]A string array of allowed tool names
-
class McpAllowedToolsMcpToolFilter: …A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
authorization: Optional[str]An OAuth access token that can be used with a remote MCP server, either with a custom MCP server URL or a service connector. Your application must handle the OAuth authorization flow and provide the token here.
-
connector_id: Optional[Literal["connector_dropbox", "connector_gmail", "connector_googlecalendar", 5 more]]Identifier for service connectors, like those available in ChatGPT. One of
server_url,connector_id, ortunnel_idmust be provided. Learn more about service connectors here.Currently supported
connector_idvalues are:-
Dropbox:
connector_dropbox -
Gmail:
connector_gmail -
Google Calendar:
connector_googlecalendar -
Google Drive:
connector_googledrive -
Microsoft Teams:
connector_microsoftteams -
Outlook Calendar:
connector_outlookcalendar -
Outlook Email:
connector_outlookemail -
SharePoint:
connector_sharepoint -
"connector_dropbox" -
"connector_gmail" -
"connector_googlecalendar" -
"connector_googledrive" -
"connector_microsoftteams" -
"connector_outlookcalendar" -
"connector_outlookemail" -
"connector_sharepoint"
-
-
defer_loading: Optional[bool]Whether this MCP tool is deferred and discovered via tool search.
-
headers: Optional[Dict[str, str]]Optional HTTP headers to send to the MCP server. Use for authentication or other purposes.
-
require_approval: Optional[McpRequireApproval]Specify which of the MCP server's tools require approval.
-
class McpRequireApprovalMcpToolApprovalFilter: …Specify which of the MCP server's tools require approval. Can be
always,never, or a filter object associated with tools that require approval.-
always: Optional[McpRequireApprovalMcpToolApprovalFilterAlways]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
never: Optional[McpRequireApprovalMcpToolApprovalFilterNever]A filter object to specify which tools are allowed.
-
read_only: Optional[bool]Indicates whether or not a tool modifies data or is read-only. If an MCP server is annotated with
readOnlyHint, it will match this filter. -
tool_names: Optional[List[str]]List of allowed tool names.
-
-
-
Literal["always", "never"]Specify a single approval policy for all tools. One of
alwaysornever. When set toalways, all tools will require approval. When set tonever, all tools will not require approval.-
"always" -
"never"
-
-
-
server_description: Optional[str]Optional description of the MCP server, used to provide more context.
-
server_url: Optional[str]The URL for the MCP server. One of
server_url,connector_id, ortunnel_idmust be provided. -
tunnel_id: Optional[str]The Secure MCP Tunnel ID to use instead of a direct server URL. One of
server_url,connector_id, ortunnel_idmust be provided.
-
-
-
tracing: Optional[RealtimeTracingConfig]Realtime API can write session traces to the Traces Dashboard. Set to null to disable tracing. Once tracing is enabled for a session, the configuration cannot be modified.
autowill create a trace for the session with default values for the workflow name, group id, and metadata.-
Literal["auto"]Enables tracing and sets default values for tracing configuration options. Always
auto."auto"
-
class TracingConfiguration: …Granular configuration for tracing.
-
group_id: Optional[str]The group id to attach to this trace to enable filtering and grouping in the Traces Dashboard.
-
metadata: Optional[object]The arbitrary metadata to attach to this trace to enable filtering in the Traces Dashboard.
-
workflow_name: Optional[str]The name of the workflow to attach to this trace. This is used to name the trace in the Traces Dashboard.
-
-
-
truncation: Optional[RealtimeTruncation]When the number of tokens in a conversation exceeds the model's input token limit, the conversation be truncated, meaning messages (starting from the oldest) will not be included in the model's context. A 32k context model with 4,096 max output tokens can only include 28,224 tokens in the context before truncation occurs.
Clients can configure truncation behavior to truncate with a lower max token limit, which is an effective way to control token usage and cost.
Truncation will reduce the number of cached tokens on the next turn (busting the cache), since messages are dropped from the beginning of the context. However, clients can also configure truncation to retain messages up to a fraction of the maximum context size, which will reduce the need for future truncations and thus improve the cache rate.
Truncation can be disabled entirely, which means the server will never truncate but would instead return an error if the conversation exceeds the model's input token limit.
-
Literal["auto", "disabled"]The truncation strategy to use for the session.
autois the default truncation strategy.disabledwill disable truncation and emit errors when the conversation exceeds the input token limit.-
"auto" -
"disabled"
-
-
class RealtimeTruncationRetentionRatio: …Retain a fraction of the conversation tokens when the conversation exceeds the input token limit. This allows you to amortize truncations across multiple turns, which can help improve cached token usage.
-
retention_ratio: floatFraction of post-instruction conversation tokens to retain (
0.0-1.0) when the conversation exceeds the input token limit. Setting this to0.8means that messages will be dropped until 80% of the maximum allowed tokens are used. This helps reduce the frequency of truncations and improve cache rates. -
type: Literal["retention_ratio"]Use retention ratio truncation.
"retention_ratio"
-
token_limits: Optional[TokenLimits]Optional custom token limits for this truncation strategy. If not provided, the model's default token limits will be used.
-
post_instructions: Optional[int]Maximum tokens allowed in the conversation after instructions (which including tool definitions). For example, setting this to 5,000 would mean that truncation would occur when the conversation exceeds 5,000 tokens after instructions. This cannot be higher than the model's context window size minus the maximum output tokens.
-
-
-
-
Returns
BinaryResponseContent
Example
import os
from openai import OpenAI
client = OpenAI(
api_key=os.environ.get("OPENAI_API_KEY"), # This is the default and can be omitted
)
call = client.realtime.calls.create(
sdp="sdp",
)
print(call)
content = call.read()
print(content)