I have run into an issue after upgrading to Asterisk 22.8.0 from 18.18.0. I also switched from chan_sip to pjsip so I’m not sure which might be the issue.
The webrtc(SIPml5) is done on Google Chrome, Safari and an Electron App and all are affected. After the upgrade video calls between two webrtc clients the video shows flawlessly but there is no audio between them. Calls where only one endpoint is webrtc seem to contain audio and audio only calls work perfectly fine between all endpoints. The call in question is originated from a call file. I have tried different combinations of codecs and there was no effect. Looking through the SDP I see that the codecs are listed properly and are the same on both sides and it contains BUNDLE audio-0 video-1. Any ideas what could be causing this? With this being such an edge case I am not sure what information is pertinent so I didn’t want to bloat this post with unneeded information but just let me know and I will add anything desired, call file, endpoint definitions, SDP, etc.