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

[Ben Langfeld]

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 at 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
>
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