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.