[SR-Users] WebRTC video calls

Daniel-Constantin Mierla miconda at gmail.com
Mon May 4 09:21:35 CEST 2015


Hello,

kamailio + rtpengine can be used for webrtc calls between browsers as
well as browser to classic sip phones. You can fine on github some
config examples, published by Carlos Ruiz Diaz.

Using this combination you can place an instance in front of asterisk
and let asterisk behave as a classic sip/rtp media server.

Cheers,
Daniel

On 02/05/15 00:44, Ivan Vujisic wrote:
> I made successful audio calls from browser to browser using Asterisk
> 13.1 and SIPML5 browser phone.
>
> Asterisk can't manage WebRTC video calls  due to lack of codec
> negotiation module, but I also faced RTP ports NAT traversal issue. To
> my understanding Kamailio is capable to resolve this.
>
> Can anybody confirm that he made successful browser to browser video
> calls with  Kamailio sip proxy / registrar in front of Asterisk PBX.
>
> Also, any link to good tutorial or doc pages will be appreciated.
>
> Best Regards,
> Ivan Vujisic
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com




More information about the sr-users mailing list