hi, when SIP user cuts the call when will livekit get to know participant disconnected?, is it instant or takes some seconds of time

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yes, i want to calculate the actual call duration, so when user cuts the call i want to know as soon as possible

Thanks — I’m responding about detecting SIP hangups as quickly as possible so you can calculate accurate call duration. Short answer: • LiveKit detects an unclean SIP disconnect based on WebRTC keep-alives and typically takes ~15–20 seconds to mark the participant disconnected and emit

participant_left

. That delay is normal to avoid false disconnects from transient network blips. Options to get the hangup time earlier (pick one or more that fit your setup): • Have your SIP gateway send an explicit event to your backend when the SIP user hangs up, and record that timestamp as the call end. Use LiveKit webhooks (`participant_joined`/`participant_left`) as a fallback to verify or reconcile later. • If your SIP gateway can call LiveKit APIs, call

RemoveParticipant

or tell the LiveKit room to

leave()

the participant immediately when SIP BYE is received — LiveKit will then emit

participant_left

immediately with a clean disconnect reason. • Add an application-level heartbeat between your SIP gateway and your backend to detect disconnects faster than the SFU timeout; when the heartbeat stops, mark the call ended. • Continue to consume

participant_left

webhooks to cover unclean cases and map disconnect reasons (

CLIENT_INITIATED

vs `CONNECTION_TIMEOUT`/`STATE_MISMATCH`) when reconciling durations. Which option would you prefer or can you implement in your SIP gateway: (A) send a direct hangup event to your backend, or (B) call LiveKit to remove the participant on SIP BYE? Tell me which and I’ll give exact implementation steps. , with interactive elements