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outbound error

Open Hollymingyi opened this issue 2 years ago • 8 comments

dear author, when i user outbound api to make call, it worked.but when i hangup the sip endpoint ,the livekit room still exist the sip endpoint ,the logs like below,

2024-01-08T16:32:31.693+0800	INFO	sip	sip/client.go:106	Updating SIP participant	{"nodeID": "NE_GA43Q49fw2GT", "roomName": "29156024", "participant": "SP_VkqWhfWLn7iD", "from": "12345", "to": "0019373086225", "address": "10.2.14.199"}
2024-01-08T16:32:31.725+0800	INFO	logger/logger.go:250	received offer for subscriber	{"nodeID": "NE_GA43Q49fw2GT"}
2024-01-08T16:32:31.770+0800	INFO	logger/logger.go:250	ICE connected	{"nodeID": "NE_GA43Q49fw2GT", "iceCandidatePair": "(local) udp4 host 172.28.0.1:40201 <-> (remote) udp4 prflx 10.2.15.102:1544 related :0"}
2024-01-08T16:32:31.779+0800	INFO	logger/logger.go:250	published track	{"nodeID": "NE_GA43Q49fw2GT", "name": "zhanghe1", "source": "MICROPHONE"}
2024-01-08T16:32:31.782+0800	INFO	logger/logger.go:250	published track	{"nodeID": "NE_GA43Q49fw2GT", "name": "zhanghe1-track", "source": "CAMERA"}
2024-01-08T16:32:31.807+0800	INFO	sip	sip/outbound.go:156	Outbound SIP update complete	{"nodeID": "NE_GA43Q49fw2GT", "roomName": "29156024", "from": "12345", "to": "0019373086225", "address": "10.2.14.199"}
2024-01-08T16:32:31.868+0800	INFO	logger/logger.go:250	received offer for subscriber	{"nodeID": "NE_GA43Q49fw2GT"}
2024-01-08T16:32:31.880+0800	INFO	logger/logger.go:250	track subscribed	{"nodeID": "NE_GA43Q49fw2GT", "participant": "admin", "track": "TR_AMPCMfLBPUvhPa", "kind": "audio"}
2024-01-08T16:32:31.900+0800	INFO	logger/logger.go:250	track subscribed	{"nodeID": "NE_GA43Q49fw2GT", "participant": "admin", "track": "TR_VC6L2ZDccZNKHf", "kind": "video"}
2024-01-08T16:32:31.952+0800	INFO	logger/logger.go:250	successfully set publisher answer	{"nodeID": "NE_GA43Q49fw2GT"}
2024-01-08T16:32:32.253+0800	INFO	logger/logger.go:250	received offer for subscriber	{"nodeID": "NE_GA43Q49fw2GT"}
2024-01-08T16:32:52.670+0800	INFO	sip	sip/inbound.go:153	BYE	{"nodeID": "NE_GA43Q49fw2GT", "tag": "t2Q81Z9UaQ7ra"}

how should i resolve them?

Hollymingyi avatar Jan 08 '24 08:01 Hollymingyi

Do I understand correctly that your issues is that the LiveKit room is not removed after SIP call ends?

Currently it won't automatically delete the LiveKit room. This is by design, because it can be done by your application using LiveKit API.

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 currently fixing.

dennwc avatar Jan 08 '24 11:01 dennwc

"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 currently fixing." yes that is exactly what my issue is , thank you,i will focus on the release log on github

Hollymingyi avatar Jan 09 '24 01:01 Hollymingyi

Call termination should now work properly for calls from SIP to LiveKit. Can you please check if it works on your end?

dennwc avatar Jan 11 '24 17:01 dennwc

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 /workspace/pkg/service/sip.go:175 github.com/livekit/protocol/livekit.(*sIPServer).serveCreateSIPParticipantProtobuf.func2 /go/pkg/mod/github.com/livekit/[email protected]/livekit/livekit_sip.twirp.go:2443 github.com/livekit/protocol/livekit.(*sIPServer).serveCreateSIPParticipantProtobuf /go/pkg/mod/github.com/livekit/[email protected]/livekit/livekit_sip.twirp.go:2444 github.com/livekit/protocol/livekit.(*sIPServer).serveCreateSIPParticipant /go/pkg/mod/github.com/livekit/[email protected]/livekit/livekit_sip.twirp.go:2305 github.com/livekit/protocol/livekit.(*sIPServer).ServeHTTP /go/pkg/mod/github.com/livekit/[email protected]/livekit/livekit_sip.twirp.go:1197 net/http.(*ServeMux).ServeHTTP /usr/local/go/src/net/http/server.go:2514 github.com/urfave/negroni/v3.(*Negroni).UseHandler.Wrap.func1 /go/pkg/mod/github.com/urfave/negroni/[email protected]/negroni.go:59 github.com/urfave/negroni/v3.HandlerFunc.ServeHTTP /go/pkg/mod/github.com/urfave/negroni/[email protected]/negroni.go:33 github.com/urfave/negroni/v3.middleware.ServeHTTP /go/pkg/mod/github.com/urfave/negroni/[email protected]/negroni.go:51

Hollymingyi avatar Jan 18 '24 09:01 Hollymingyi

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?

Hollymingyi avatar Jan 18 '24 10:01 Hollymingyi

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

Hollymingyi avatar Jan 18 '24 10:01 Hollymingyi

same problem, are there any updates?

vlxdisluv avatar Feb 14 '25 11:02 vlxdisluv

@Hollymingyi try changing the SIP trunk provider. In our case, the issue was that some SIP trunk providers don't send any signals indicating that the call has ended.

vlxdisluv avatar Feb 18 '25 06:02 vlxdisluv