Streaming WebSocket API

Authenticate with a temporary token via query parameter. Use this where request headers cannot be set — for example, browser WebSocket connections — and prefer the Authorization header everywhere else. Generate the temporary token server-side; never expose your permanent API key in a URL.

The speech model to use for the session. See Select the speech model for the differences between models.

Steers transcription toward a set of languages by biasing output toward them on a per-token basis while still allowing native code-switching among them. Pass a list of language codes for the languages you expect (for example, ["en", "es"]), or a single-element list (for example, ["es"]) for a monolingual session. When unset, no steering is applied and the model code-switches natively across all of its supported languages. This is distinct from language_detection, which only controls whether the detected language is reported. Accepted codes: en, es, fr, de, it, pt, tr, nl, sv, no, da, fi, hi, vi, ar, he, ja, zh. Universal-3.5 Pro Streaming only.

Whether to return language_code and language_confidence in turn messages. Available on Universal-3.5 Pro Streaming (which natively code-switches across all of its supported languages by default) and on Universal Streaming Multilingual only.

Enable domain-specific transcription models to improve accuracy for specialized terminology. Set to "medical-v1" to enable Medical Mode for improved accuracy of medical terms such as medications, procedures, conditions, and dosages. Supported languages: English (en), Spanish (es), German (de), French (fr). If used with an unsupported language, the parameter is ignored and a warning is returned.

Latency and accuracy preset that controls the model's turn-detection and partial-emission defaults. max_accuracy favors transcription quality, min_latency favors speed, and balanced trades off between the two. When omitted, the server applies its own preset, which determines the defaults for the mode-dependent fields min_turn_silence and interruption_delay. Universal-3.5 Pro Streaming only.

Encoding of the audio stream. pcm_s16le and pcm_mulaw are raw PCM formats. opus is raw Opus packets, where each binary WebSocket message must contain exactly one Opus packet. ogg_opus is an Ogg-encapsulated Opus byte stream (the format produced by ffmpeg, gstreamer, opusenc, and browser MediaRecorder); binary WebSocket messages can be arbitrary chunks of the stream. For both Opus encodings, sample_rate is ignored — the stream is self-describing.

Sample rate of the audio stream in Hz. Accepts any integer from 8000 to 96000. Ignored for Opus encodings (opus, ogg_opus).

Silence duration in milliseconds before a speculative end-of-turn check. On Universal-3.5 Pro Streaming the check is punctuation-based (if terminal punctuation is found the turn ends, otherwise a partial is emitted and the turn continues) and the default is mode-dependent, set by the mode preset. On Universal Streaming the check is confidence-based and the default is 400 ms. Clamped to the range 50 to 10000 ms. See Configuring Turn Detection for configuration details.

Maximum silence in milliseconds before the turn is forced to end, regardless of punctuation. Defaults are 1536 ms on Universal-3.5 Pro Streaming (768 ms when speaker_labels is enabled) and 1280 ms on Universal Streaming. See Configuring Turn Detection for configuration details.

The confidence threshold (0.0 to 1.0) to use when determining if the end of a turn has been reached. See Turn Detection for configuration details. Universal Streaming (English and Multilingual) only.

The confidence threshold (0.0 to 1.0) for classifying audio frames as silence. Frames with VAD confidence below this value are considered silent. Increase for noisy environments to reduce false speech detection. Defaults are 0.2 on Universal-3.5 Pro Streaming (0.5 when speaker_labels is enabled) and 0.4 on Universal Streaming.

How soon the first partial is emitted in milliseconds. Accepts a value from 0 to 1000. Useful for tuning voice agent barge-in responsiveness or allowing earlier partials for early LLM inference. Larger values are more confident on interruptions, smaller values result in faster time to first partial. The server adds a fixed 256 ms on top of this value, so 0 yields an effective delay of 256 ms and 500 yields 756 ms. The default is mode-dependent and set by the server based on the mode preset. Universal-3.5 Pro Streaming only.

Whether to emit additional partial transcripts during long turns at a steady ~3 second cadence. When enabled (default), additional partials covering the full turn transcript are emitted approximately every 3 seconds while speech continues. When disabled, only one early partial is emitted near turn start. The first partial (timed by interruption_delay) is unaffected. When speaker_labels is enabled the server disables continuous partials by default. Universal-3.5 Pro Streaming only.

Whether to emit partial transcripts during the turn. When enabled (default), partial transcripts are forwarded as speech is still in progress alongside final turns. When disabled, only final turns (with end_of_turn true) are sent. Defaults to false when redact_pii is enabled, to prevent unredacted partial transcripts from reaching the client; set explicitly to true to override.

Whether to return formatted final transcripts. Universal Streaming (English and Multilingual) only.

A contextual prompt describing what the audio is about, such as its domain, scenario, or full conversation details. The model uses this context to better recognize the vocabulary it makes likely, for example priming medical terminology for a cardiology call. It carries context about your audio, not transcription instructions, so formatting or behavioral commands (such as punctuation rules) are not supported. Maximum 1750 characters. Universal-3.5 Pro Streaming only. See Prompting and keyterms for details.

A list of words and phrases to improve recognition accuracy for. Maximum 100 terms. See Keyterms Prompting for more details.

Your voice agent's spoken text (TTS reply). The model uses this as context for the next user turn, which improves accuracy on short or ambiguous replies and on spelled-out entities like emails or IDs. Set at connection time to seed the model with your agent's opening greeting, and/or update mid-stream via UpdateConfiguration after each agent reply. Each UpdateConfiguration replaces the previously set value. Maximum 1750 characters per value. Universal-3.5 Pro Streaming only. See Context Carryover.

Advanced. Maximum number of prior conversation entries (user transcripts and any agent_context values) carried forward as context for each transcription. Range 0 to 100. Set to 0 to disable automatic context carryover entirely. When unset, the server default applies (currently 5 turns). Most integrations should leave this unset. Universal-3.5 Pro Streaming only.

Whether to enable Streaming Speaker Diarization. When enabled, each Turn event will include a speaker_label field and each final word in the words array will include a speaker field for word-level speaker attribution.

A hard cap on the number of speaker labels in the audio stream (integer, 1-10). This is a strict limit, not a hint — once it is reached, any additional speakers are merged into the closest existing label rather than given a new one. Give the model a little headroom above the number of speakers you expect; setting it too high can cause over-splitting and return more speakers than are actually present. Only used when speaker_labels is enabled. See Streaming Diarization for more details.

Enable Voice Focus to isolate the primary voice and suppress background noise before transcription. Set to near-field for close-talking microphones (for example, headsets or phones) or far-field for distant microphones (for example, conference rooms). When unset, Voice Focus is off.

Controls how aggressively Voice Focus suppresses background audio, from 0.0 (least) to 1.0 (most). Higher values are more aggressive. Requires voice_focus to be set, otherwise a validation error is returned.

Redact PII from the transcribed text using the Redact PII model, can be true or false. Only applies to final turns. See PII Redaction for more details.

The list of PII Redaction policies to enable. Requires redact_pii to be true. See PII redaction for more details.

The replacement logic for detected PII, can be entity_name or hash. Requires redact_pii to be true. See PII redaction for more details.

Filter profanity from the transcribed text, can be true or false. See Profanity Filtering for more details.

JSON-stringified LLM Gateway configuration that processes each finalized turn. Follows the same interface as the Chat Completions endpoint and accepts model, messages, tools, tool_choice, post_processing_steps, and max_tokens. See Apply LLM Gateway to Streaming for the full schema and examples.

Optional time in seconds of inactivity before session is terminated (integer, minimum 5, maximum 3600). If not set, no inactivity timeout is applied.