Card #509 · OpenAI Platform & APIs
SIP Realtime API drops final transcription events on caller hangup
When a caller hangs up immediately after speaking in a SIP Realtime session, the WebSocket closes before final transcription events are delivered. This results in the final utterance being missing from the conversation history, despite being present in the call recording.
Pain score
0.17/ 1.00 · weighted product of four components
Reach0.40distinct authors (log-normalized)
Recency1.00freshness decay, half-life months
Engagement1.00upvotes + comments, normalized
Monetization0.50willingness-to-pay cues in evidence
§ Evidence — 3 verified quotes
If the caller ends the call after it’s connected, OpenAI immediately closes the session WebSocket. This prevents pending events (tool calls, transcription, output buffering) from being delivered, resulting in incomplete tool calls and partial transcriptions.
final transcription is not guaranteed when someone talks continuously and hangs up immediately.
Because no delta or completed event is delivered, the final caller utterance cannot be persisted in our conversation history.
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