[SR-Users] Kamailio as a SIP softphone to WebRTC/SIP-over-websockets proxy?

[Daniel-Constantin Mierla]

Hello,

Kamailio can be used to gateway signaling from webrtc/websocket to other
transports like UDP/TCP/TLS. See websocket module for an example to
start with and search of github for more advanced example by Carlos Ruiz
Diaz. Some talks were given at Kamailio World Conference in the past
that shows further examples.

If you need to do gatewaying of the SRTP to RTP, then you have to add
RTPEngine as well.

Cheers,
Daniel

On 28/02/15 22:13, Rion Carter wrote:
> I'm pretty new to SIP, RTP/SRTP, WebRTC and Websockets, so I hope this
> question is coherent. I have a group of SIP Softphones that need to
> connect to a WebRTC/SIP-over-Websockets server. Can Kamailio be
> configured to let me do this?
>
> Any examples, tutorials or documentation would be appreciated. I'm
> trying to determine how feasible this task is. :)
>
> Thanks!
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20150302/5bd44dd6/attachment.html>