[SR-Users] RTCWeb Breaker question

Daniel-Constantin Mierla miconda at gmail.com
Wed Feb 10 21:57:49 CET 2016


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).


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
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Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com

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