[SR-Users] WebRTC + Kamailio as SIP-proxy + rtpproxy(?) as media-proxy

[Camille Oudot]

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