First I want to give Denys a huge shout-out for all of the help he has given me.  It is wonderful that boards like this exists and people are so willing to help a newbie learn.   I am on what I am hoping is my last major issue with WebRTCóWebRTC calls (using tryit-jssip Chrome or Firefox).   I am using Kamailio 5, and Asterisk 15 (pjsip). I am making calls between two WebRTC clients  - Client1, and Client2 (using tryit-jssip)   Problem:  If Client1 calls Client2, and Client2  ‘ANSWERS’, I only have audio/video on Client1.  Client2 gets no audio/video, but is connected.  If I switch things up and call Client1 from Client2, the same thing happens (Client2 has audio/video and Client1 does not); I can only get audio/video on the calling laptop; the called laptop has no audio/video, but is connected.  I see no errors in any of the logs.   I am hoping that someone out there has seen this behavior before and has an idea as to the cause and possible solution.