[SR-Users] RTCWeb Breaker question

Daniel-Constantin Mierla miconda at gmail.com
Wed Feb 10 22:19:39 CET 2016


RTPBreaker as per wiki link was never intended to be a transcoder.

Anyhow, you need a media server here - I know that FreeSwitch did a lot
of video work lately, so it would be the first option I would look at.

Also, if you find a classic sip video transcoder, you can use
kamailio+rtpengine to decrypt/encrypt the leg to webrtc and get sip and
plain rtp to this transcoder.

Cheers,
Daniel

On 10/02/16 22:12, SamyGo wrote:
>
> Thanks for clarification Daniel. That obviously mean that I can not
> achieve transcoding (VP8/H264).
>
> Given my objective do you have any recommendations ?
>
> Thanks for your valuable time..
> Regards,
> Sammy
>
> On Feb 10, 2016 15:58, "Daniel-Constantin Mierla" <miconda at gmail.com
> <mailto:miconda at gmail.com>> wrote:
>
>     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 at lists.sip-router.org <mailto: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://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
>     Book: SIP Routing With Kamailio - http://www.asipto.com
>     http://miconda.eu
>
>
>     _______________________________________________
>     SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>     list
>     sr-users at lists.sip-router.org <mailto: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
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160210/dd4b9a3e/attachment.html>


More information about the sr-users mailing list