<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></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 .<br class=""><br class="">the problems that i faced with this is the following and i hope i could get an advise here.<br class=""><br 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> <<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 :).<br class=""><br class="">i have two questions and i hope you could give me some advise. <br class=""><br 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.<br class=""><br 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? <br class="">1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call.<br class=""><br 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).<br class=""><br class="">2.1) asterisk is muggling the audio and video streams ? <br class=""><br class="">This is good information for all guys out there that wants to support video on webrtc in asterisk 13<br class=""><br class="">Thanks<br class=""><br class="">Javier Riveros</body></html>