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
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