[SR-Users] Transcoding

[SamyGo]

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 at evaristesys.com>
wrote:

> Richard,
>
> That's very exciting news!
>
> On January 26, 2018 10:44:51 AM EST, Richard Fuchs <rfuchs at 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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