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.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users