We haven't seen too many issues from SIP endpoints. Where we see the problem is from participants coming from the browser, just as the OP mentioned above. Are there ways to test/monitor the WebRTC connections to see if that is indeed where the bot...
So there is no way to change that presentation setting that Cisco has locked? I wonder if there is a command in the API that you can change that setting. Has anyone else tried this?
Are you referring to the callLegProfile settings for qualityPresentation in the API? What about if you set the presentation to "unrestricted" or "max1080p30"?
