The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams 🙂 it just mark video packets not touch anything else and web browser show video on web page now I’m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :).
i have two questions and i hope you could give me some advise.
1) after marking video packet I’m able to make Dial() between two webrtc peers but i get one way audio and video on callee party, “after 3 minutes on call” i get two way audio and video on all parties seems to be not just a problem on a missing keyframe.
1.1) the 3 minutes delay only happen using chrome stable , could be a dtls problem when asterisk make an offer to other endpoint?
1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call.
2) after marking video packets i realize that when you make a call with video and you involve on dialplan an application like playback or music on hold any application that played audio files (audio and video never work).
2.1) asterisk is muggling the audio and video streams ?
This is good information for all guys out there that wants to support video on webrtc in asterisk 13