Synthesize Speech (WebSocket)

You open a persistent WebSocket connection and send text messages. The server streams audio chunks back over the same connection — no per-request overhead, no repeated handshakes. This gives you the lowest possible latency.

Best for voice agents and interactive applications that send multiple synthesis requests in a session, where avoiding connection setup on every call makes a measurable difference.

Message Pipelining

Messages on a connection are processed in order, so you don't need to wait for the contextCreated acknowledgment before sending text. For the lowest latency, send each message as soon as it is ready — waiting for the acknowledgment adds a full network round trip before the first audio chunk arrives. If you omit contextId, messages are automatically routed to the connection's auto-created context.

Text Limits

Each send_text message accepts up to 2,000 characters. Text accumulates on the server and is synthesized when the buffer flushes — on an explicit flush_context, when max_buffer_delay_ms elapses, or once accumulated text reaches buffer_char_threshold (1,000 characters by default, configurable up to 2,000).

If you have a single block of text rather than a stream of them, the Streaming API takes up to 4,000 characters in one request.

Timestamp Transport Strategy

When using timestamp alignment, you can choose how timestamps are delivered alongside audio using timestampTransportStrategy:

• SYNC (default): Each chunk contains both audio and timestamps together.
• ASYNC: Audio chunks arrive first, with timestamps following in separate trailing messages. This reduces time-to-first-audio.

See Timestamps for details on how each mode works.

Code Examples

API Reference

Next Steps