I’m not sure whether my question is more related to jssip or to webrtc, but i’m desperate so i’ll try here.
I have a door bell, which can initiate sip video calls with ulaw/h264. I’ve installed asterisk and configured it to accept sip endpoints through udp and websocket. Using examples from the internet i’ve built an app to receive calls using react-native-webrtc and react-native-jssip. It even works if i place an audio call from asterisk to my RN app. I can answer it, audio travels both directions - everything is perfect:
LOG event newRTCSession fired
LOG incoming call
LOG Answering call
LOG event session.peerconnection fired
LOG event session.connection.negotiationneeded fired
LOG event session.connection.signalingstatechange fired
LOG event session.connection.addstream fired
LOG {"_reactTag": “4cdeddb6-cf47-4dac-88bf-736234d874ca”, “_tracks”: [{"_constraints": [Object], “_enabled”: true, “_settings”: [Object], “id”: “aaff823f-eaf6-4819-af79-de060e76e3d8”, “kind”: “audio”, “label”: “Audio”, “muted”: false, “readyState”: “live”, “remote”: true}], “active”: true, “id”: “2f7e0d37-538f-45bd-9f1b-bb837090af40”}
LOG event session.connecting fired
LOG CONNECT
LOG event session.connection.signalingstatechange fired
LOG event session.connection.iceconnectionstatechange fired
LOG event session.connection.icegatheringstatechange fired
LOG event session.connection.connectionstatechange fired
LOG event session.connection.icecandidate fired
LOG event session.connection.icecandidate fired
LOG event session.connection.icegatheringstatechange fired
LOG event session.accepted fired
LOG event session.confirmed fired
LOG event session.connection.connectionstatechange fired
LOG event session.connection.iceconnectionstatechange fired
However, if i enable h264 codec on asterisk for RN endpoint - i got an error from webrtc:
LOG event newRTCSession fired
LOG incoming call
LOG Answering call
LOG event session.peerconnection fired
LOG event session.connection.negotiationneeded fired
LOG event session.connection.signalingstatechange fired
LOG event session.failed fired
LOG {“cause”: “WebRTC Error”, “message”: null, “originator”: “system”}
LOG event session.connection.iceconnectionstatechange fired
I googled and tryed everything before writing this question, maybe someone can guide me - most likely i’m doing something wrong or missing some important bit.
Is there any debug for webrtc similar to jssip’s JsSIP.debug.enable(‘JsSIP:*’); ?
I would really appreciate any advice,
thank you in advance!