Hollymingyi

ok,i will capture the rtp package and check the timestamp

dear author, sometimes when i published video mediatrack,it rendered very soon,some times it rendered need about 1m. but the log of livekit is normal.

hello zhao, because i use that profile for volte call into livekit. then, yes,it did work when i add the 102 profile in livekit codecs.

"On the other hand, the SIP participant in that room should disappear after call hangup. If it doesn't happen, then you probably hit a issue with BYE that we are...

> termination sh which release get rid of it? the newest release have error like below: ERROR livekit service/sip.go:199 cannot update sip participant {"error": "request timed out"} github.com/livekit/livekit-server/pkg/service.(*SIPService).updateParticipant /workspace/pkg/service/sip.go:199 github.com/livekit/livekit-server/pkg/service.(*SIPService).CreateSIPParticipant...

> Call termination should now work properly for calls from SIP to LiveKit. Can you please check if it works on your end? did you mean terminate the call manually?

when i ended the sip call by hangup the sip phone, it still exist in the room. which commit should i checkout to resolve this bug

dear david， docker hub pull latest，1.2.1 and 1.1.0 ，i tried them all，cant solve the problem. also，could u tell me how to sovle the latency about rtmp egress and egress，now when...

hi sean, what i want is that when i call into livekit room via freeswitch , i can communicate with livekit endpoint with video and audio together ,but now only...

ah ,if use ingress and egress to connect freeswitch and livekit , for example ,rtmp protocol ,the latency is more than 9s,its not suitable for our applications. right now ,...