[SR-Users] RTCWeb Breaker question

[SamyGo]

Hi Again,
That is really interesting, I'd like to know how since we do have our own
transcoding mechanism inside some MCU server and I might extract that and
engage it using RTPengine.

Thanks for the idea.

Regards,
Sammy.


On Wed, Feb 10, 2016 at 4:19 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> 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>
> 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 listsr-users at 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 at lists.sip-router.org
>> http://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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160210/a7acaad5/attachment.html>