2014-12-18 20:38 GMT+02:00 Andrey Utkin <andrey.krieger.ut...@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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
