SIP to WebRTC Proxy / Gateway

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SIP to WebRTC Proxy / Gateway

Rion Carter
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!

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Re: 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 <[hidden email]> 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
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Saúl Ibarra Corretgé
AG Projects




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Re: SIP to WebRTC Proxy / Gateway

Eric Tamme-3
Re: DTLS-SRTP + ICE -> plain RTP, this can be done using opensips with rtpengine, I have tested it using oversip -> opensips + rtpengine -> legacy sip.

But generally speaking if you are "pretty new to SIP, RTP and WebRTC" you are likely going to run into a lot of head aches with WebRTC and "legacy" SIP device interop.  I don't mean to be a downer ... thats just the way things are really.


On 03/04/2015 02:35 AM, Saúl Ibarra Corretgé wrote:
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 [hidden email] 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
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Saúl Ibarra Corretgé
AG Projects





_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users