Skip to main content
Use streaming decisions to evaluate audio as it arrives. Your application sends audio chunks over a WebSocket connection and receives answers with probabilities throughout the stream. See Stream decisions for the full message reference.
Tazo Small is currently unavailable through the API, including streaming. Requests return model_unavailable. For assistance, contact support@tazolabs.com. Use tazo-large for API requests.

​
Start a session

Open a connection to:
Send session.start within five seconds of connecting. Include your API key, the context, and one question:
The server returns session.ready with the accepted configuration and limits. Wait for this event before sending audio.

​
Send audio

Declare the source format in session.start.input_audio, then send the matching raw mono samples as binary WebSocket messages: channels must be 1. The default remains PCM16 at 16 kHz. A telephone stream using μ-law declares {"encoding":"mulaw","sample_rate_hz":8000,"channels":1}. Read the format from your provider’s configuration or stream metadata; raw samples do not contain enough information to discover it. Unwrap provider JSON/base64 or RTP before forwarding the audio payload. File containers and codecs such as Opus and AAC are not supported here. Unsupported declarations return unsupported_audio_format; a new session is required to change the format. Messages must contain complete samples and at most 200 ms of source audio. Read session.ready.max_audio_message_bytes: 3,200 bytes for 8 kHz PCM16, 1,600 for G.711, 6,400 for 16 kHz PCM16, or 19,200 for 48 kHz PCM16. Send only samples, without WAV headers or base64 text. A 100 ms message is recommended. The server decodes and resamples to mono 16 kHz PCM16 before evaluation, retaining filter state across messages and flushing the final samples at commit/finish. Resampling adds buffering delay; decision offsets use the converted sample clock. When a turn limit applies, it counts all incoming audio, including audio held by the filter. Existing 16 kHz PCM16 is passed through unchanged. The default audio_scope is chunk: each evaluation uses new audio since the previous evaluation and can stream continuously without periodic commits. Set audio_scope to turn explicitly to evaluate all audio received in the current turn. Set evaluation_interval_ms to a value from 250 to 30,000 ms to choose the evaluation interval. The default is 1,000 ms. Responses may arrive less often when an evaluation takes longer. The server may shorten the accepted interval to fit the model’s input limit; read session.ready.evaluation_interval_ms. Each evaluation counts against the account’s rate limit. The example uses 1,000 ms, which can use 60 of the account’s 500 requests per rolling minute for that model. The default 1,000 ms interval can use 60 requests per minute per continuous stream. Allow room for other streams and uploaded decisions.

​
Receive decisions

Listen for decision.updated events while continuing to send audio. For example, a request about the weather can produce:
answers uses the question and option IDs from your request. audio_start_ms and audio_end_ms identify the evaluated audio range within the current turn. Your application decides how to act on each result.

​
Complete a turn

A turn is a group of audio chunks that your application submits for evaluation. To complete it, stop sending audio and send:
The server evaluates any remaining audio and returns decision.finalized. In chunk mode, this final answer covers the last chunk. Keep earlier updates if your application needs results from the rest of the turn. Wait for decision.finalized before sending the next turn on the same connection. Buffer any incoming audio during this wait. To end the session, send {"type": "input.finish"}; the server finishes pending audio and sends session.closed before closing the connection. In chunk mode, session.ready.max_turn_audio_ms is null: there is no total audio-duration cap and no need to commit periodically. The server releases audio after handing it to inference and keeps each evaluation within the model’s input limit. Call input.commit when you want a turn boundary, or input.finish when done. The playground keeps its own 30-second recording limit. In turn mode, session.ready.max_turn_audio_ms is a number reflecting the model and server limits. Commit before exceeding it. Send audio at real-time speed in either mode; audio_buffer_full means pending audio exceeded the server’s buffer capacity. See Limits. If the server sends an error event, read error.code and error.message. When fatal is true, the connection closes.

​
Audio credits

When usage billing is enabled, one WebSocket session is one billing operation, including all turns. Tazo adds received samples across the session and rounds the total up once to a whole second, storing the deduction in milliseconds. A 100 ms stream costs 1 second; ten 100 ms chunks in one stream also cost 1 second. Intermediate evaluations do not create extra usage charges. See Credits and audio usage for exhaustion, disconnect, and shared balance behavior.