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Re: DTLS-SRTP + ICE -> plain RTP, this can be done using opensips
with rtpengine, I have tested it using oversip -> opensips +
rtpengine -> legacy sip.<br>
<br>
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.<br>
<br>
<br>
<div class="moz-cite-prefix">On 03/04/2015 02:35 AM, Saúl Ibarra
Corretgé wrote:<br>
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cite="mid:402A87DA-2C2C-4B68-AAC6-DB64327985A2@ag-projects.com"
type="cite">
<pre wrap="">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 <a class="moz-txt-link-rfc2396E" href="mailto:rion.carter@gmail.com"><rion.carter@gmail.com></a> wrote:
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<pre wrap="">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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<pre wrap="">
--
Saúl Ibarra Corretgé
AG Projects
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