Hi All,We are experiencing issues with our video conference system. When a user joins from a specific network, their video and audio transmission become stuck after some time. Upon investigation using the webrtc-internals tool in Google Chrome, we observed that no bytes were received. However, the peer connection remains active, and we noticed some packet sending within the same connection's inbound-rtp session.
Interestingly, if the user changes their network and refreshes the application, the system works fine. We have also implemented TURN and STUN servers.
The problem seems to lie in the video and audio data not being received, despite the peer connection functioning properly. Any insights on this issue would be greatly appreciated.