[OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[Olle E. Johansson]

Asterisk 11 has some early support for SIP over websockets, but that's far from being compatible with WebRTC. The standards for WebRTC are still evolving and require much more. It's a good step forward, but the ASterisk team is not there yet... :-)

SIP over websockets is currently a draft that has progressed nicely through the IETF process.
http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-05

/O

31 okt 2012 kl. 17:05 skrev Bogdan-Andrei Iancu <bogdan at opensips.org>:

> Hi guys,
> 
> Thanks for this - I will take a look at this websocket to see what about - is there any RFC or similar ?
> 
> Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> On 10/31/2012 03:53 PM, Ali Pey wrote:
>> 
>> Hi Bogdan,
>> 
>> 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.
>> 
>> 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.
>> 
>> Regards,
>> Ali Pey
>> 
>> On Wed, Oct 31, 2012 at 8:21 AM, Saúl Ibarra Corretgé <saul at ag-projects.com> wrote:
>> 
>> On Oct 31, 2012, at 12:52 PM, Bogdan-Andrei Iancu wrote:
>> 
>> > Hi Saul,
>> >
>> > 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 ?
>> >
>> 
>> 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.
>> 
>> 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 https://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412 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.
>> 
>> 
>> Regards,
>> 
>> --
>> Saúl Ibarra Corretgé
>> AG Projects
>> 
>> 
>> 
>> 
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