Re: [SR-Users] RTCWeb Breaker question

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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users