Thanks for the response. You're right, the media stream is making it all the way back to my PC, I just don't hear anything. And yes, my speakers are turned up...
I'm not sure what to try next... On Fri, Dec 19, 2014 at 12:31 PM, Richard Fuchs <rfu...@sipwise.com> wrote: > > On 12/19/14 10:47, Marc Soda wrote: > > I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not > > getting audio back to my browser. From a packet capture I can see media > > from the browser to rtpengine, and then bi-directional RTP back and > > forth from my asterisk server, but rtpengine is not sending the media on > > to the browser, i.e.: > > > > browser ---------> kamailio/rtpengine <---------> asterisk > > > > This is the output from rtpengine: > > > > https://gist.github.com/marcantonio/bfe72644306b205cc7e1 > > You've caught the same thing as Juha did just earlier, Firefox is doing > something new called Trickle ICE, which at the moment breaks > communications with endpoints not supporting it (such as rtpengine). > > The second call you posted seems fine. The error you're seeing is > because RTP was received before DTLS was established and so is expected. > You can try --dtls-passive as a possible fix. Media should start to flow > after DTLS gets established though, and according to the logs, media was > indeed seen in both directions. Try tcpdump to confirm. > > cheers > > _______________________________________________ > 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 >
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