[asterisk-users] PJSIP - Video Support for WebRTC
mjordan at digium.com
Mon Mar 23 11:34:10 CDT 2015
On Mon, Mar 23, 2015 at 8:55 AM, Gosmac <goseeped at 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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