I am making a video call from webrtc to sip client using the sip plugin. At first, both parties see the videos without any problems, but after a while, the SIP client’s video drops on the webrtc client. When I get the Pcap file and examine it, I see that there is packet loss in the video packets coming from the SIP server to Janus. In the first packet loss, a mute event from Janus comes to the webrtc client. However, even though the package continues to come back, the unmute event does not come to the webrtc client. That’s why the video never comes back after it drops. Can this be solved?
Log on the console in Webrtc client:
janus.js:1966 Remote track muted: Event {isTrusted: true, type: 'mute', target: MediaStreamTrack, currentTarget: MediaStreamTrack, eventPhase: 2, …}
janus.js:1969 Removing remote track
siptest.js:529 Remote track (mid=1) removed: MediaStreamTrack {kind: 'video', id: '3b2d8b66-baf2-4ebc-8840-aed04cfbb380', label: '3b2d8b66-baf2-4ebc-8840-aed04cfbb380', enabled: true, muted: true, …}