we are testing a webrtc installation using SIPml5 and asterisk 16.9.0.
We can correctly connect to our pbx using sipml5 and do audio calls with no issues, everything works, so far.
The problem happens when we try to do a video call. The user that starts the call will have is microphone not working but if the same user tries an audio only call or receives a video call, his microphone will be working correctly. We tried on different browsers, Chrome, Edge, Safari using different OS, Windows 10, MacOS, Debian, Fedora but everyone has this issue and we can’t figure out what it is.
Our pjsip.conf file contains:
[bob] type=aor max_contacts=1 [bob] type=auth auth_type=userpass username=bob password=********* [bob] type=endpoint context=default direct_media=no ;allow=!all,ulaw,vp8,h264 allow=opus,alaw,h264 aors=bob auth=bob max_audio_streams=10 max_video_streams=10 webrtc=yes rtcp_mux = yes [lucy] type=aor max_contacts=1 [lucy] type=auth auth_type=userpass username=lucy password=******** [lucy] type=endpoint context=default direct_media=no ;allow=!all,ulaw,vp8,h264 allow=opus,alaw,h264 aors=lucy auth=lucy max_audio_streams=10 max_video_streams=10 webrtc=yes rtcp_mux = yes
Our rtp.conf file contains
[general] rtpstart=10000 rtpend=20000 stunaddr=stun.l.google.com:19302
If you need more let me know and i’ll provide it. This is the first time we are approaching webrtc and we didn’t find a lot of topics on the internet with those same issues as we are having