Thanks for reaching out! I understand you're looking for a way to know exactly when bidirectional audio starts flowing between participants in LiveKit. While there isn't a specific webhook event that signals this moment directly, there are some key events that can really help you follow the call lifecycle: • participant_joined: This happens when a participant joins a room. • track_published: This is triggered when a participant's media track, like audio, is published to the room. It's the closest indicator that audio is available and flowing. • sip_call_started: This event shows when a SIP call has started. For inbound SIP calls, usually the flow goes like this: the SIP call is accepted, a participant joins the room (which triggers participant_joined), and then the audio track gets published (triggering track_published). So, the track_published event is probably your best sign that audio is actively flowing. If you'd like me to walk you through this further or if you have a specific scenario in mind, please don’t hesitate to ask. I’m here to help! Sources: Webhooks | LiveKit Docs | livekit-rtc/livekit/rtc/_proto/ffi_pb2.py | pkg/sip/inbound.go