Production Patterns

Use these patterns when running the standalone WebSocket STT endpoint in production.

Connection Recovery

Treat the WebSocket session as disposable. Reconnect on network failure, server close, idle timeout, and process restart.

Event	Action
Connection fails before `open`	Retry with backoff. Do not send audio until the connection is open.
Connection closes unexpectedly	Stop sending audio, preserve buffered audio, reconnect, then resume streaming.
Error message received	Log `errors[].code`, `errors[].title`, and `errors[].source.parameter`. Reconnect only after fixing parameter errors.
Graceful shutdown	Send `{"type": "CloseStream"}` and wait for final transcripts before closing the socket.

Set all query parameters on every reconnect. STT configuration cannot be changed mid-session.

Backoff

Use bounded exponential backoff with jitter.

Attempt	Base delay
1	250 ms
2	500 ms
3	1 s
4	2 s
5+	5 s max

Add random jitter of 0-500 ms per attempt. Reset the attempt counter after a stable connection. Do not retry immediately on authentication or validation errors. Fix the API key, query parameters, engine, model, or format first.

Partials

Enable interim_results=true when the application needs live captions or low-latency UI updates.

Message	Handling
`is_final: false`	Display as temporary text. Replace it when a newer partial arrives. Do not persist it as final transcript.
`is_final: true`	Commit to the transcript. Do not replace it with later partials.
`utterance_end: true`	Treat as a segment boundary. Do not render an empty transcript as text.

Store final transcript segments separately from the current partial. This prevents duplicate text when a final result arrives after one or more interim results.

Audio Buffering

Buffer audio at the producer boundary, not inside the WebSocket send loop.

Control	Recommendation
Chunk size	Send 2048-8192 byte binary frames.
Queue size	Set a maximum buffered duration, such as 5-10 seconds.
Backpressure	Pause or drop low-priority audio when the queue is full.
Reconnect	Keep a short rolling buffer only if retranscription after reconnect is required.

Avoid unbounded queues. A slow or disconnected socket should not grow memory usage indefinitely. For live audio, prefer dropping stale buffered audio over sending it late. Late audio increases transcript delay and can make captions appear out of sync.

Keepalive

For Deepgram sessions, send {"type": "KeepAlive"} during long silence periods. Keep sending audio as binary frames when audio is available. For other engines, use the WebSocket client’s ping/pong support when available and reconnect on missed heartbeats.

Monitoring

Track connection, latency, transcript, and buffer metrics.

Metric	Purpose
Connection attempts	Detect retry loops and regional network issues.
Connection duration	Detect unstable sessions and idle timeout patterns.
Close code and reason	Separate expected closes from failures.
Error codes	Identify invalid parameters and engine compatibility issues.
Audio queue depth	Detect send-loop backpressure.
Partial-to-final latency	Measure caption freshness.
Final transcript count	Detect stalled recognition.
Empty final count	Detect silence segmentation behavior.

Log the selected transcription_engine, model, input_format, sample_rate, and interim_results value with each session. Redact API keys and user audio.

Shutdown

Use graceful shutdown for planned stops.

Drain the audio queue.
Send {"type": "CloseStream"}.
Wait for final transcript messages.
Close the WebSocket.

Set a shutdown timeout. If final messages do not arrive before the timeout, close the socket and mark the transcript as incomplete.

WebSocket

REST API

In-Call

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Production Patterns

Connection Recovery

Backoff

Partials

Audio Buffering

Keepalive

Monitoring

Shutdown

​Connection Recovery

​Backoff

​Partials

​Audio Buffering

​Keepalive

​Monitoring

​Shutdown

Connection Recovery

Backoff

Partials

Audio Buffering

Keepalive

Monitoring

Shutdown