[OpenSIPS-Users] SIP to WebRTC Proxy / Gateway

[Eric Tamme]

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 <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!
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>> Users at lists.opensips.org
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> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
>
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