Hello,
if you use one of latest asterisk version as media server, it should
have support for webrtc media handling, so just forward the calls to it.
For a media gateway, you can use rtpproxy enginge module and application
along with kamailio (for stable version 4.1, rtpproxy-ng module).
Cheers,
Daniel
On 30/04/14 23:43, Patrik Kristel wrote:
Hello,
I have Kamailio with websocket module and I want to connect Asterisk
as media server. I'm trying to route calls from web JsSIP users to
non-web users I would like to ask how can I implement it? Can i use
the rtpproxy-ng Module and here setup IP of Asterisk? Or is there any
other way to do it?
Thank you for your help!
Regards,
Patrik Kristel
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -
http://www.asipto.com
http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda