<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">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 .<div class=""><br class=""></div><div class="">the problems that i faced with this is the following and i hope i could get an advise here.</div><div class=""><br class=""></div><div class="">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 <a href="http://tryit.jssip.net/" class="">http://tryit.jssip.net/</a> is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :).</div><div class=""><br class=""></div><div class="">i have two questions and i hope you could give me some advise. </div><div class=""><br class=""></div><div class="">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.</div><div class=""><br class=""></div><div class=""> 1.1) the 3 minutes delay only happen using chrome stable , could be a dtls problem when asterisk make an offer to other endpoint? </div><div class=""> 1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call.</div><div class=""><br class=""></div><div class="">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).</div><div class=""> </div><div class="">2.1) asterisk is muggling the audio and video streams ? </div><div class=""><br class=""></div><div class="">This is good information for all guys out there that wants to support video on webrtc in asterisk 13</div><div class=""><br class=""></div><div class="">Javier Riveros</div></body></html>