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.