dencommHub — Developer Portal

Connection

Field	Value
Endpoint	wss://api.denthub.ai/stt/realtime/stream
Query	?key=<key>
Audio	PCM16LE · 16000 Hz · mono · binary frames
Max conns/pod	200 (HTTP 503 when exceeded)

start↑ first frame (text)

Field	Type	Required	Value
type	string	yes	"start"
sample_rate	int	yes	16000
codec	string	yes	"pcm_s16le"
channels	int	yes	1

audio↑ binary frames

Send raw PCM16LE bytes as binary WebSocket frames — no WAV header, no base64, no JSON wrapping. Any frame size works; send continuously, without gating on silence yourself.

The gateway re-frames your stream internally into 32 ms VAD windows and detects utterance boundaries server-side, so you never need to run your own voice-activity detection.

end↑ graceful close (text)

Send this text frame to flush any pending utterance and close cleanly. After sending it, you may close the WebSocket yourself, or wait for the server to close it once flushing finishes.

{ "type": "end" }

result↓ one per utterance

{
  "type": "result",
  "utt_index": 0,
  "session_id": "b1f7…",
  "abs_start_utc": "2026-07-10T05:12:01.80Z",
  "abs_end_utc":   "2026-07-10T05:12:05.20Z",
  "duration_ms": 3400,
  "text": "Patient presents with sensitivity on the upper right quadrant.",
  "segments": [
    { "start_ms": 800, "end_ms": 4200,
      "text": "Patient presents with sensitivity…", "language": "en" }
  ],
  "words": [ { "word": "Patient", "start_ms": 800, "end_ms": 1180 }, … ],
  "status": "ok",
  "forced_cut": false,
  "continues": false
}

Field	Type	Description
utt_index	int	0-based, monotonic per connection
session_id	uuid	Connection session ID
abs_start_utc / abs_end_utc	string (ISO 8601)	Absolute utterance start/end (UTC)
duration_ms	int	Utterance duration (sample-derived)
text	string	Full transcription text
segments[]	array	start_ms / end_ms / text / language, recording-relative
words[]	array	word / start_ms / end_ms, recording-relative
status	string	"ok" · "empty" · "failed"
forced_cut	bool	true if the utterance was force-split at max length
continues	bool	true if the next utt_index continues this utterance

Every timestamp here — abs_*_utc, and each start_ms/end_ms inside segments[] and words[] — is recording-relative, derived from a sample-count clock, not message-arrival time. Use these to line results up against the audio you sent, even under network jitter.

↓ session · warning · error

{"type":"session","event":"started","session_id":"…"}   // started · ended
{"type":"warning","code":"SLOW_CONSUMER"}                  // backpressure
{"type":"utterance_dropped","utt_index":8,"code":"STT_OVERLOAD"}
{"type":"error","utt_index":9,"code":"STT_TIMEOUT","message":"…","retryable":true}

session — connection lifecycle: started and ended.
warning — backpressure signal (SLOW_CONSUMER): you're sending audio faster than the gateway can process; slow down or check your network.
utterance_dropped — the STT backend was overloaded and this utterance was skipped; the gap in utt_index tells you which one.
error — a per-utterance failure; check retryable to decide whether to wait or move on.

Close codes

Code	Meaning
1000	Normal close, after you sent end
1001	Going away — server is draining; reconnect to a new session
1011	Internal error — log it and reconnect