27 Nov
2014
27 Nov
'14
3:04 p.m.
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users