I'm trying to use jssip and it's sample client (tryit.jssip.net
) to make video calls through asterisk. Everything works fine if i initiate calls from client to asterisk. However there is a problem if i try to receive video call on sample client. After answer it says "no remote video", although if i check webrtc-internal in chrome, it displays graphs in section "Stats graphs for RTCInboundRTPVideoStream_938134447 (inbound-rtp)" - that remote video stream is successfully received by Chrome.
Funny thing that if i tap hold and unhold in tryit sample client, remote video appears after that.
I've compared SDPs of successful (tryit.jssip -> asterisk) and problematic (asterisk -> tryit.jssip) the only difference was in the naming of media sections. Jssip names section 0 1, while asterisk names it audio-0 and video-1. I assumed that there may be a problem in jssip processing those section names, so i changed res/res_pjsip_session.c in asterisk to simplify section naming, but that didn't help.
Any idea how to make tryit.jssip to display remote video immediately after answer without toggling hold/unhold?
Thank you very much in advance.