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<tt>Hi guys,<br>
<br>
Thanks for this - I will take a look at this websocket to see what
about - is there any RFC or similar ?<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 10/31/2012 03:53 PM, Ali Pey wrote:
<blockquote
cite="mid:CA+q4kR+80_=wgzymDLoTY4y0sK2Ypz5QTLpq2zgfBq3gVRj2Jg@mail.gmail.com"
type="cite">Hi Bogdan,
<div><br>
</div>
<div>Saul is correct. The key thing is to support WebSocket
transport. WebRTC is becoming quite popular and seems to be the
thing of future. It is already added in asterisk (version 11
released last week) and it is being added to Kamalio. Using
OverSIP and the OpenSIPS would make things just more complex
specially for larger deployments.</div>
<div><br>
</div>
<div>Also, deployments that are already using opensips would want
to provide WebRTC based apps and it makes a lot of sense to
natively support it with opensips. Asterisk's implementation
seemed quite complete. Maybe you can have a look at that.</div>
<div><br>
</div>
<div>Regards,</div>
<div>Ali Pey<br>
<br>
<div class="gmail_quote">On Wed, Oct 31, 2012 at 8:21 AM, Saúl
Ibarra Corretgé <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:saul@ag-projects.com" target="_blank">saul@ag-projects.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div class="im"><br>
On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote:<br>
<br>
> Hi Saul,<br>
><br>
> OK, aside the TCP part (which anyhow is scheduled for
fixing) and some extra parsing, does supporting WebRTC
imply something more on the OpenSIPS side ?<br>
><br>
<br>
</div>
It requires that OpenSIPS is able to use SIP over a
WebSocket transport. So OpenSIPS would need support for
WebSocket. The transport behaves roughly the same as TCP,
with the difference that you may only get a single SIP
packet in each WebSocket segment. There are other couple of
minor things, but the core of it is supporting the WebSocket
transport.<br>
<br>
I didn't have the time to test OpenSIPS behind a OverSIP
instance, to verify if there is anything to be fixed
beforehand, but according to <a moz-do-not-send="true"
href="https://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412"
target="_blank">https://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412</a>
Via parsing will fail because if doesn't recognize ws and
wss as valid transports. I had a look at the attached patch
and it seems to solve the problem by accepting any Via
transport parameter. I guess that's a good idea, so OpenSIPS
would not care about any transport used in the path, as long
as it doesn't need to use it, but I guess some checks would
need to be done to validate if the topmost Via has a
transport OpenSIPS understands. Not sure if this check is
already done though.<br>
<div class="HOEnZb">
<div class="h5"><br>
<br>
Regards,<br>
<br>
--<br>
Saúl Ibarra Corretgé<br>
AG Projects<br>
<br>
<br>
<br>
<br>
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</blockquote>
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