[SR-Users] webrtc clients support using rtpengine

[Andrey Utkin]

2014-12-18 20:38 GMT+02:00 Andrey Utkin <andrey.krieger.utkin at gmail.com>:
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
> This doesn't work: few seconds after answer, there's no media from
> remote subscriber, then linphone android agent hangs up; sipml hangs
> pretending being in call.
> rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
> kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
> ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde

Doesn't work in different way:
jssip web agent calls to android linphone: call is connected, but
android doesn't see peer's video, and jssip doesn't have peer's audio:
rtpengine: https://gist.github.com/krieger-od/cd63ef99d06212b30379
kamailio: https://gist.github.com/krieger-od/55c4b5785c5821315955
ngrep: https://gist.github.com/krieger-od/650575c0f96285d78d17

-- 
Andrey Utkin