Hi Steve,
In the past I've used WebRTC2SIP gateway project from doubango : https://www.doubango.org/webrtc2sip/
I was able to do transcoding for video calls as well. As far as I recall it was CPU intensive and may require some WebRTC(HTTP/HTTPS) loadbalancing to be able to handle an satisfactory amount of calls.
In my scenario I had to transcode VP8/VP9<==>H264 streams.

I hope we get better( & efficient) video-transcoding projects soon. 

Regards,
Sammy


On Fri, Jan 26, 2018 at 10:48 AM, Alex Balashov <abalashov@evaristesys.com> wrote:
Richard,

That's very exciting news!

On January 26, 2018 10:44:51 AM EST, Richard Fuchs <rfuchs@sipwise.com> wrote:
>On 2018-01-26 08:57 AM, Wilkins, Steve wrote:
>>
>> Hello All,
>>
>> I am currently using Kamailio and Asterisk on Centos 7 servers and
>> trying to enable WebRTC jsSIP clients to be able to do Audio/Video
>> calls with Provider Phones (Purple, Z, Sorenson, etc.…), however, the
>
>> providers do not have vp8 codecs (which is what the WebRTC clients
>use
>> for Audio) so I believe I will need a media proxy server to resolve
>> the video issues.  My question is, can rtpproxy or rtpengine perform
>> this transcoding? If so, and if rtpengine is the way to go, should I
>> use Ubuntu for the rtpengine since it is the only one that seems to
>> have a working installation?
>>
>
>Work on transcoding support for rtpengine is currently underway.
>However, the initial focus will be on audio codecs only. Video support
>might be added in the future.
>
>Cheers


-- Alex

--
Sent via mobile, please forgive typos and brevity.

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