Realtime SIP caller hangup still drops final input transcription events

Hi Mark G.,

Yes, I am aware of the recent SIP-related issues. However, the problem I reported has been occurring consistently both before and after those fixes were rolled out.

I believe the root cause is that when connecting to the Realtime API via SIP, the WebSocket session closes immediately as soon as the caller hangs up. Because of this, the final transcription events, such as conversation.item.input_audio_transcription.delta or conversation.item.input_audio_transcription.completed, are never sent.