27 Nov
2014
27 Nov
'14
2:45 p.m.
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