Realtime SIP caller hangup still drops final input transcription events

@Toshi, your read is likely right: in SIP Realtime sessions, the WebSocket can close fast after BYE / hangup. If VAD hasn’t committed the final audio yet, the transcription events may never fire.

Right now, final transcription is not guaranteed when someone talks continuously and hangs up immediately.

A few things worth logging:

• speech_started
• speech_stopped
• input_audio_buffer.committed

If you see speech_started but no committed before disconnect, that points to the final audio never becoming a conversation item.

Best workaround for now is to save transcription deltas as they arrive, and use the Twilio recording as fallback for the caller’s last utterance in these quick-hangup cases.

-Mark G.