On Mon, Mar 23, 2015 at 8:55 AM, Gosmac <gosee...@gmail.com> wrote: > Hey i have an interesting topic to discuss here. > > 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 >
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