I have run into an issue after switching from chan_sip to pjsip. With pjsip my video calls between webrtc clients do not contain audio. I have tested with multiple Asterisk versions of 20 22 and 23 but I get the same results with every one after chan_sip was deprecated.
The webrtc(SIPml5) is done on Google Chrome and an Electron App and the same thing happens whether it is Electron to Electron, Chrome to Chrome or Electron to Chrome. Audio only calls work perfectly fine. I have tried different combinations of codecs and there was no effect. This is inside of a NAT and with RTP set debug on I can see that the rtp is going to the right place. Using chrome://webrtc-internals it looks like the audio stream is there but my SIPml5 code doesn’t seem to recognize it. Does anyone know of something that changed in regards to how the stream would be recognized in chan_sip vs pjsip? I have noticed certain changes to the SDP from the switch to pjsip but nothing specific stands out as an obvious issue. I saw in a previous post that someone was having webrtc audio issues recently so I tried what worked for them which was changing to using bundle=no. To actually get that to take effect I had to replace webrtc=yes to include all of the options it is supposed to be a shortcut for but the call completely fails in that scenario. I have attached the cli output with debugging enabled, 18 being the working call and 23 being nonworking. Is there any other difference between chan_sip and pjsip that could possibly be causing this issue?
18_pbx_video.txt (22.0 KB)
23_pbx_video.txt (19.0 KB)