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
Thanks.
Even stranger, I get a media stream back to the browser when I use Chrome (the first was with Firefox), but I still hear nothing. Also I get errors like this in the log:
SRTP output wanted, but no crypto suite was negotiated Full output:
https://gist.github.com/marcantonio/6c5414aa931a8f1c0072
On Fri, Dec 19, 2014 at 10:47 AM, Marc Soda msoda@coredial.com 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
Thanks.
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:
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
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 rfuchs@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:
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
On Fri, Dec 19, 2014 at 12:31 PM, Richard Fuchs rfuchs@sipwise.com wrote:
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).
I figured out the issue. It was totally client related.
Are there plans for rtpengine support trickle ICE anytime soon? My (limited) understand is that without it I can't support Firefox WebRTC clients.
Thanks, Marc
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Marc Soda -
Richard Fuchs