Realtime SIP caller hangup still drops final input transcription events

Hi Mark G., Thanks for the info. I’ll proceed with the implementation based on your advice. If possible, it would be great to have this information documented in the API or SDK docs. It’s a tricky issue that’s hard to notice until you’ve been running in production for a while, so having a heads-up would help many developers avoid the same pitfall. I really appreciate all your support so far. You can go ahead and close this ticket.