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resources/realtime/subresources/transcription_sessions/index.md 2026-06-10 15:48 UTC to 2026-06-12 00:01 UTC

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Transcription Sessions

Create transcription session

post /realtime/transcription_sessions

Create an ephemeral API token for use in client-side applications with the Realtime API specifically for realtime transcriptions. Can be configured with the same session parameters as the transcription_session.update client event.

It responds with a session object, plus a client_secret key which contains a usable ephemeral API token that can be used to authenticate browser clients for the Realtime API.

Returns the created Realtime transcription session object, plus an ephemeral key.

Body Parameters

  • include: optional array of "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 "pcm16" or "g711_ulaw" or "g711_alaw"

    The format of input audio. Options are pcm16, g711_ulaw, or g711_alaw. For pcm16, 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 object { type }

    Configuration for input audio noise reduction. This can be set to null to 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_field is for close-talking microphones such as headphones, far_field is 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 "minimal" or "low" or "medium" or 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-whisper in GA Realtime sessions.

      • "minimal"

      • "low"

      • "medium"

      • "high"

      • "xhigh"

    • language: optional string

      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 string or "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 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, and gpt-realtime-whisper. Use gpt-4o-transcribe-diarize when you need diarization with speaker labels.

      • string

      • "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 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, and gpt-realtime-whisper. Use gpt-4o-transcribe-diarize when 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 string

      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. For gpt-4o-transcribe models (excluding gpt-4o-transcribe-diarize), the prompt is a free text string, for example "expect words related to technology". Prompt is not supported with gpt-realtime-whisper in GA Realtime sessions.

  • turn_detection: optional object { prefix_padding_ms, silence_duration_ms, threshold, type }

    Configuration for turn detection. Can be set to null to 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 number

      Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.

    • silence_duration_ms: optional number

      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 number

      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 "server_vad"

      Type of turn detection. Only server_vad is currently supported for transcription sessions.

      • "server_vad"

Returns

  • client_secret: object { expires_at, value }

    Ephemeral key returned by the API. Only present when the session is created on the server via REST API.

    • expires_at: number

      Timestamp for when the token expires. Currently, all tokens expire after one minute.

    • value: string

      Ephemeral 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 string

    The format of input audio. Options are pcm16, g711_ulaw, or g711_alaw.

  • input_audio_transcription: optional object { language, model, prompt }

    Configuration of the transcription model.

    • language: optional string

      The language of the input audio.

    • model: optional string or "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 3 more

      The model used 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, and gpt-realtime-whisper.

      • string

      • "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 3 more

        The model used 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, and gpt-realtime-whisper.

        • "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 string

      The prompt configured for input audio transcription, when present.

  • modalities: optional array of "text" or "audio"

    The set of modalities the model can respond with. To disable audio, set this to ["text"].

    • "text"

    • "audio"

  • turn_detection: optional object { prefix_padding_ms, silence_duration_ms, threshold, type }

    Configuration for turn detection. Can be set to null to 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 number

      Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.

    • silence_duration_ms: optional number

      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 number

      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 string

      Type of turn detection, only server_vad is currently supported.

Example

curl https://api.openai.com/v1/realtime/transcription_sessions \
    -H 'Content-Type: application/json' \
    -H "Authorization: Bearer $OPENAI_API_KEY" \
    -d '{}'

Response

{
  "client_secret": {
    "expires_at": 0,
    "value": "value"
  },
  "input_audio_format": "input_audio_format",
  "input_audio_transcription": {
    "language": "language",
    "model": "whisper-1",
    "prompt": "prompt"
  },
  "modalities": [
    "text"
  ],
  "turn_detection": {
    "prefix_padding_ms": 0,
    "silence_duration_ms": 0,
    "threshold": 0,
    "type": "type"
  }
}

Example

curl -X POST https://api.openai.com/v1/realtime/transcription_sessions \
  -H "Authorization: Bearer $OPENAI_API_KEY" \
  -H "Content-Type: application/json" \
  -d '{}'

Response

{
  "id": "sess_BBwZc7cFV3XizEyKGDCGL",
  "object": "realtime.transcription_session",
  "modalities": ["audio", "text"],
  "turn_detection": {
    "type": "server_vad",
    "threshold": 0.5,
    "prefix_padding_ms": 300,
    "silence_duration_ms": 200
  },
  "input_audio_format": "pcm16",
  "input_audio_transcription": {
    "model": "gpt-4o-transcribe",
    "language": null,
    "prompt": ""
  },
  "client_secret": null
}

Domain Types

Transcription Session Create Response

  • TranscriptionSessionCreateResponse object { client_secret, input_audio_format, input_audio_transcription, 2 more }

    A 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: object { expires_at, value }

      Ephemeral key returned by the API. Only present when the session is created on the server via REST API.

      • expires_at: number

        Timestamp for when the token expires. Currently, all tokens expire after one minute.

      • value: string

        Ephemeral 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 string

      The format of input audio. Options are pcm16, g711_ulaw, or g711_alaw.

    • input_audio_transcription: optional object { language, model, prompt }

      Configuration of the transcription model.

      • language: optional string

        The language of the input audio.

      • model: optional string or "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 3 more

        The model used 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, and gpt-realtime-whisper.

        • string

        • "whisper-1" or "gpt-4o-mini-transcribe" or "gpt-4o-mini-transcribe-2025-12-15" or 3 more

          The model used 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, and gpt-realtime-whisper.

          • "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 string

        The prompt configured for input audio transcription, when present.

    • modalities: optional array of "text" or "audio"

      The set of modalities the model can respond with. To disable audio, set this to ["text"].

      • "text"

      • "audio"

    • turn_detection: optional object { prefix_padding_ms, silence_duration_ms, threshold, type }

      Configuration for turn detection. Can be set to null to 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 number

        Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.

      • silence_duration_ms: optional number

        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 number

        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 string

        Type of turn detection, only server_vad is currently supported.