# "WebRTC Error" on incoming jssip call with video

**URL:** <https://react-native-webrtc.discourse.group/t/webrtc-error-on-incoming-jssip-call-with-video/1371>\
**Category:** Help\
**Created:** [October 2, 2021, 3:11pm UTC](https://react-native-webrtc.discourse.group/t/webrtc-error-on-incoming-jssip-call-with-video/1371 "2021-10-02T15:11:11Z")\
**Posts on this page:** 1\
**Page:** 1

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**Author:** ![serg](https://avatars.discourse-cdn.com/v4/letter/s/ecd19e/32.png) [@serg](https://react-native-webrtc.discourse.group/u/serg)\
**Post date:** [October 2, 2021, 3:11pm UTC](https://react-native-webrtc.discourse.group/t/webrtc-error-on-incoming-jssip-call-with-video/1371/1 "2021-10-02T15:11:11Z")

</div>

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!