Try using the chrome://webrtc-internals tools to check what the browser sees. It will contain graphs on how many packets are being decoded, whether it detects resolution, fps, etc. You can also try recording the call in the SIP plugin to the Janus MJR format, that you can then convert using janus-pp-rec, to see if the frames are decodable/playable in a regular video player.
I did capture call using janus in mjr format and converted to playable video file.
peer-video.webm (is black I am not able to play).
user-video.webm (is playable).
I also did capture same session using tcpdump in janus. Here is mjr, video and tcpdump also. Download zip
I am also attaching picture of webrtc internals from firefox from different call where I get image distorted:
