Create transcription session
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.
Request body
modalitiesobject
The set of modalities the model can respond with. To disable audio, set this to ["text"].input_audio_formatstringDefaults: pcm16
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.pcm16string
g711_ulawstring
g711_alawstring
input_audio_transcriptionobject
Configuration for input audio transcription. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service.modelstring
The model to use for transcription, current options are
gpt-4o-transcribe,gpt-4o-mini-transcribe, andwhisper-1.gpt-4o-transcribestring
gpt-4o-mini-transcribestring
whisper-1string
languagestring
The language of the input audio. Supplying the input language in ISO-639-1 (e.g.
en) format will improve accuracy and latency.promptstring
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, the prompt is a free text string, for example "expect words related to technology".
turn_detectionobject
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 conjuction 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.typestringDefaults: server_vad
Type of turn detection.server_vadstring
semantic_vadstring
eagernessstringDefaults: 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.lowstring
mediumstring
highstring
autostring
thresholdnumber
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.prefix_padding_msinteger
Used only for
server_vadmode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms.silence_duration_msinteger
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.create_responsebooleanDefaults: true
Whether or not to automatically generate a response when a VAD stop event occurs. Not available for transcription sessions.interrupt_responsebooleanDefaults: true
Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e.
conversationofauto) when a VAD start event occurs. Not available for transcription sessions.
input_audio_noise_reductionobjectDefaults: null
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.typestring
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_fieldstring
far_fieldstring
includearray
The set of items to include in the transcription. Current available items are:
item.input_audio_transcription.logprobs
itemsstring
Response
The created Realtime transcription session object, plus an ephemeral key
1 curl -X POST https://api.openai.com/v1/realtime/transcription_sessions \2 -H "Authorization: Bearer $OPENAI_API_KEY" \3 -H "Content-Type: application/json" \4 -d '{}'
1 {2 "id": "sess_BBwZc7cFV3XizEyKGDCGL",3 "object": "realtime.transcription_session",4 "modalities": ["audio", "text"],5 "turn_detection": {6 "type": "server_vad",7 "threshold": 0.5,8 "prefix_padding_ms": 300,9 "silence_duration_ms": 20010 },11 "input_audio_format": "pcm16",12 "input_audio_transcription": {13 "model": "gpt-4o-transcribe",14 "language": null,15 "prompt": ""16 },17 "client_secret": null18 }