Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Juha Heinanen
Richard Fuchs writes: > In the third case, the audio stream was setup as a=sendonly, explaining > the one-way audio. Probably caused by Firefox not being able to access > the playback device. as i reported in another message, i have also lately noticed in my tests that firefox sends invite with a

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs
On 12/18/14 13:38, Andrey Utkin wrote: > 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 > > >

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
Also I encounter such issue: even in the working scenario, the hangup of one peer doesn't make the call end for another peer. rtpengine: https://gist.github.com/krieger-od/1cfe84b53dc0d29cfb90 kamailio: https://gist.github.com/krieger-od/11c6bbf7dad15382e81b ngrep: https://gist.github.com/krieger-o

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
2014-12-18 20:38 GMT+02:00 Andrey Utkin : > 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 > >

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
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

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
2014-12-18 20:09 GMT+02:00 Andrey Utkin : > 2014-12-18 20:05 GMT+02:00 Richard Fuchs : >> Amazon NAT is exactly why I've mentioned it, because on an Amazon >> system, if you don't use the --interface option correctly >> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors >> that

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
2014-12-18 20:05 GMT+02:00 Richard Fuchs : > Amazon NAT is exactly why I've mentioned it, because on an Amazon > system, if you don't use the --interface option correctly > ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors > that show in your log. Thank you, will try with suc

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs
On 12/18/14 12:55, Andrey Utkin wrote: > 2014-12-18 19:30 GMT+02:00 Richard Fuchs : >> Write error on RTP socket usually indicates an incorrect network setup, >> for example trying to use a source address for IP packets which isn't >> bound to any local network interface (especially if you're sitti

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin
2014-12-18 19:30 GMT+02:00 Richard Fuchs : > Write error on RTP socket usually indicates an incorrect network setup, > for example trying to use a source address for IP packets which isn't > bound to any local network interface (especially if you're sitting > behind NAT), or local iptables rules re

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs
On 12/18/14 12:11, Andrey Utkin wrote: > Hi! > I need to establish calls between WebRTC and usual SIP clients > (exactly, sipml/jssip and linphone-android). > I used configs from https://github.com/caruizdiaz/kamailio-ws and > latest git master HEAD of both kamailio and > rtpengine. I got calls fro