rapid-honey-43838
02/27/2026, 3:46 PMtall-airline-16756
02/27/2026, 3:47 PMtall-airline-16756
02/27/2026, 3:47 PMparticipant_joined — when a SIP caller joins a room via the trunk
• participant_left — when the SIP caller disconnects
• room_started / room_finished — when a room begins or ends due to SIP activity
• track_published / track_unpublished — when audio from the SIP participant becomes available or stops
• egress_started / egress_ended — if recording/egress is configured for the SIP call
Webhooks are delivered as HTTP POSTs to your configured endpoint and include participant metadata that identifies the SIP connection so your backend can react in real time.
If you want, I can:
• show a sample webhook payload for a SIP participant,
• explain how to distinguish SIP participants from normal WebRTC participants in the payload, or
• help you troubleshoot missing webhooks (what endpoint to check, common failure reasons). Which would be most helpful? Sources: sip | livekit/livekit_composite | DeepWikirapid-honey-43838
02/27/2026, 3:47 PMtall-airline-16756
02/27/2026, 3:48 PMparticipant_joined + track_published to infer answered).
• Help troubleshoot missing call webhooks (what to check on your webhook endpoint and common causes).
Which option should I expand on, or tell me the specific call events you need (e.g., ringing, answered, busy, failed, transfer).