[OpenSIPS-Users] SIP to WebRTC Proxy / Gateway

[Saúl Ibarra Corretgé]

IIRC OpenSIPS hasn’t landed WebSocket support yet, but you could use OverSIP for translating SIP over UDP/TCP/TLS to WebSocket, acting as an outbound proxy, for example.

Nevertheless, if you are connecting to a WebRTC gateway thing, signalling is likely the least of your worries, because your soft phone probably doesn’t support the media capabilities mandated by WebRTC.


Cheers,

On 28 Feb 2015, at 22:00, Rion Carter <rion.carter at gmail.com> wrote:

> I'm pretty new to SIP, RTP and WebRTC. I am in need of a gateway or proxy that can let me use an existing SIP Soft-phone to connect to a WebRTC/SIP-over-websockets server (the  WebRTC/SIP-over-websockets server does not provide a way for regular SIP softphones to connect).
> 
> Would OpenSIPS be able to proxy my requests from my softphone to the WebRTC endpoint? I have examined the documentation and if I've missed something I apologize. Most everything I read emphasizes connecting webrtc clients to a server, and my need is different than that.
> 
> Any examples, tutorials or documentation would be appreciated.
> 
> Thanks!
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Saúl Ibarra Corretgé
AG Projects



-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 842 bytes
Desc: Message signed with OpenPGP using GPGMail
URL: <http://lists.opensips.org/pipermail/users/attachments/20150304/c0247331/attachment.pgp>