Hello,
I'm new to WebRTC although I've been using kamailio as sip proxy server for few months now. What I really do not know and trying to understand is -
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to plain UAC/WebRTC based UAC ?
b) What to use for media proxying (this really baffles me..) rtpproxy or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation between them anywhere?
c) I am not behind NAT and do not want secure web-sockets, so any sample config I can refer to ?
d) Most likely, I'd be dealing with WebRTC <----->kamailio <-------> Freeswitch, but any pointers for WebRTC UAC to WebRTC based UAC or normal UAC would really be helpful.
Kindly accept my thanks in advance for this !!
Hi,
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to plain UAC/WebRTC based UAC ?
Yes, kamailio can do SIP over websocket, so all you need is a javascript SIP stack (e.g. JsSIP, jain-sip JS, ...) on your WebRTC enabled client.
b) What to use for media proxying (this really baffles me..) rtpproxy or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation between them anywhere?
you will need to be able to translate WebRTC RTP (RTP/SAVPF) to other RTP profiles like RTP/AVP. Only rtpengine can do this (note that mediaproxy-ng is the old name for rtpengine).
c) I am not behind NAT and do not want secure web-sockets, so any sample config I can refer to ?
If you familiar with kamailio cfg scripting you can try to start something from scratch (building a simple proxy is quite straightforward). Otherwise i don't know any example file that does all you need.
See examples/websocket.cfg for websocket handling. You can disable the registrar and the NAT stuff in it if you don't need them.
Cheers,
The main thing you need to look out for is that your registrar supports the Path and Outbound specifications in order to correctly route INVITEs to your WebSocket clients via the edge proxy. I'm in a situation right now where I'm having some difficulty getting a Kamailio WebSocket edge proxy playing nice with an Asterisk 1.4 registrar, which doesn't support those specs. If anyone has any tips, I'd love to hear them.
On 27 November 2014 at 10:45, Camille Oudot camille.oudot@orange.com wrote:
Hi,
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to plain UAC/WebRTC based UAC ?
Yes, kamailio can do SIP over websocket, so all you need is a javascript SIP stack (e.g. JsSIP, jain-sip JS, ...) on your WebRTC enabled client.
b) What to use for media proxying (this really baffles me..) rtpproxy or rtpengine (?) or mediaproxy or rtpproxy-ng ? Is there any relation between them anywhere?
you will need to be able to translate WebRTC RTP (RTP/SAVPF) to other RTP profiles like RTP/AVP. Only rtpengine can do this (note that mediaproxy-ng is the old name for rtpengine).
c) I am not behind NAT and do not want secure web-sockets, so any sample config I can refer to ?
If you familiar with kamailio cfg scripting you can try to start something from scratch (building a simple proxy is quite straightforward). Otherwise i don't know any example file that does all you need.
See examples/websocket.cfg for websocket handling. You can disable the registrar and the NAT stuff in it if you don't need them.
Cheers,
-- Camille
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Ben Langfeld -
Camille Oudot -
Rahul MathuR