Hello, I think that page was created when RTPEngine was at the beginning with WebRTC features. Right now it should just work to use Kamailio+RTPEngine to communicate with classic SIP phone, given that there is no need to transcode (encryption/decryption is done by RTPEngine, as well as de-multiplexing streams).
Cheers, Daniel On 10/02/16 20:49, SamyGo wrote: > Hi All, > > reference to this > link: https://www.kamailio.org/wiki/devel/rtcweb_breaker#scenarios > > I want to know if the module to communicate with RTCWeb Breaker is > available or it was just a proposal and no more under consideration. > > I have webrtc clients registered to Kamailio but due to lack of > (scalable/efficient) transcoding capabilities they can not make video > calls to Video IP-Phones. > > I tried using webrtc2sip from doubango telecom and it actually enabled > me to achieve the goal, the problem with that case is webrtc2sip is > working with sipml5 client and there is not a big list of WebRTC > clients that work with it. > > If I can achieve the referred rtc_web_breaker architecture then I > believe a lot of webRTC clients will be able to integrate with my setup. > > Thanks, > > Regards, > Sammy > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
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