> ## Documentation Index > Fetch the complete documentation index at: https://openai-hd4n6.mintlify.site/llms.txt > Use this file to discover all available pages before exploring further. # 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. ## OpenAPI ````yaml api-definition.yaml post /realtime/transcription_sessions openapi: 3.0.0 info: title: OpenAI API description: >- The OpenAI REST API. Please see https://platform.openai.com/docs/api-reference for more details. version: 2.3.0 termsOfService: https://openai.com/policies/terms-of-use contact: name: OpenAI Support url: https://help.openai.com/ license: name: MIT url: https://github.com/openai/openai-openapi/blob/master/LICENSE servers: - url: https://api.openai.com/v1 security: - ApiKeyAuth: [] tags: - name: Assistants description: Build Assistants that can call models and use tools. - name: Audio description: Turn audio into text or text into audio. - name: Chat description: >- Given a list of messages comprising a conversation, the model will return a response. - name: Completions description: >- Given a prompt, the model will return one or more predicted completions, and can also return the probabilities of alternative tokens at each position. - name: Embeddings description: >- Get a vector representation of a given input that can be easily consumed by machine learning models and algorithms. - name: Evals description: Manage and run evals in the OpenAI platform. - name: Fine-tuning description: Manage fine-tuning jobs to tailor a model to your specific training data. - name: Batch description: Create large batches of API requests to run asynchronously. - name: Files description: >- Files are used to upload documents that can be used with features like Assistants and Fine-tuning. - name: Uploads description: Use Uploads to upload large files in multiple parts. - name: Images description: Given a prompt and/or an input image, the model will generate a new image. - name: Models description: List and describe the various models available in the API. - name: Moderations description: >- Given text and/or image inputs, classifies if those inputs are potentially harmful. - name: Audit Logs description: List user actions and configuration changes within this organization. paths: /realtime/transcription_sessions: post: tags: - Realtime summary: Create transcription session description: > 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. operationId: create-realtime-transcription-session requestBody: description: Create an ephemeral API key with the given session configuration. required: true content: application/json: schema: $ref: '#/components/schemas/RealtimeTranscriptionSessionCreateRequest' responses: '200': description: Session created successfully. content: application/json: schema: $ref: >- #/components/schemas/RealtimeTranscriptionSessionCreateResponse components: schemas: RealtimeTranscriptionSessionCreateRequest: type: object description: Realtime transcription session object configuration. properties: modalities: description: | The set of modalities the model can respond with. To disable audio, set this to ["text"]. items: type: string default: - text - audio enum: - text - audio input_audio_format: type: string default: pcm16 enum: - pcm16 - g711_ulaw - g711_alaw description: > 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. input_audio_transcription: type: object description: > Configuration for input audio transcription. The client can optionally set the language and prompt for transcription, these offer additional guidance to the transcription service. properties: model: type: string description: > The model to use for transcription, current options are `gpt-4o-transcribe`, `gpt-4o-mini-transcribe`, and `whisper-1`. enum: - gpt-4o-transcribe - gpt-4o-mini-transcribe - whisper-1 language: type: string description: > The language of the input audio. Supplying the input language in [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`) format will improve accuracy and latency. prompt: type: string description: > 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](/docs/guides/speech-to-text#prompting). For `gpt-4o-transcribe` models, the prompt is a free text string, for example "expect words related to technology". turn_detection: type: object description: > Configuration for turn detection, ether Server VAD or Semantic VAD. This can be set to `null` to 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. properties: type: type: string default: server_vad enum: - server_vad - semantic_vad description: | Type of turn detection. eagerness: type: string default: auto enum: - low - medium - high - auto description: > Used only for `semantic_vad` mode. The eagerness of the model to respond. `low` will wait longer for the user to continue speaking, `high` will respond more quickly. `auto` is the default and is equivalent to `medium`. threshold: type: number description: > Used only for `server_vad` mode. 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_ms: type: integer description: > Used only for `server_vad` mode. Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. silence_duration_ms: type: integer description: > Used only for `server_vad` mode. 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_response: type: boolean default: true description: > Whether or not to automatically generate a response when a VAD stop event occurs. Not available for transcription sessions. interrupt_response: type: boolean default: true description: > Whether or not to automatically interrupt any ongoing response with output to the default conversation (i.e. `conversation` of `auto`) when a VAD start event occurs. Not available for transcription sessions. input_audio_noise_reduction: type: object default: null description: > 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. properties: type: type: string enum: - near_field - far_field description: > 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. include: type: array items: type: string description: > The set of items to include in the transcription. Current available items are: - `item.input_audio_transcription.logprobs` RealtimeTranscriptionSessionCreateResponse: type: object description: > 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 one minute. This property is not present when a session is updated via the WebSocket API. properties: client_secret: type: object description: | Ephemeral key returned by the API. Only present when the session is created on the server via REST API. properties: value: type: string description: > 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. expires_at: type: integer description: > Timestamp for when the token expires. Currently, all tokens expire after one minute. required: - value - expires_at modalities: description: | The set of modalities the model can respond with. To disable audio, set this to ["text"]. items: type: string enum: - text - audio input_audio_format: type: string description: > The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`. input_audio_transcription: type: object description: | Configuration of the transcription model. properties: model: type: string description: > The model to use for transcription. Can be `gpt-4o-transcribe`, `gpt-4o-mini-transcribe`, or `whisper-1`. enum: - gpt-4o-transcribe - gpt-4o-mini-transcribe - whisper-1 language: type: string description: > The language of the input audio. Supplying the input language in [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`) format will improve accuracy and latency. prompt: type: string description: > An optional text to guide the model's style or continue a previous audio segment. The [prompt](/docs/guides/speech-to-text#prompting) should match the audio language. turn_detection: type: object description: > 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. properties: type: type: string description: > Type of turn detection, only `server_vad` is currently supported. threshold: type: number description: > 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_ms: type: integer description: | Amount of audio to include before the VAD detected speech (in milliseconds). Defaults to 300ms. silence_duration_ms: type: integer description: > 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. required: - client_secret x-oaiMeta: name: The transcription session object group: realtime example: | { "id": "sess_BBwZc7cFV3XizEyKGDCGL", "object": "realtime.transcription_session", "expires_at": 1742188264, "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 } securitySchemes: ApiKeyAuth: type: http scheme: bearer ````