Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking

On Aug 2, 2013, at 12:30 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:

> Hi Adrian,
>  
> >I would suggest that best practice is for the SIP traffic to be routed to the SIP Proxy/Registrar/Presence Agent of the given domain behind the gateway. That element has more proper capabilities for handling forking or any other SIP features. 
>  
> Ok, but how are you going to ensure that such element exists?

The gateway knows, it queries the DNS for the domain SIP service records. If no proper records exist, the call flow will stop as there is nothing to rote the calls flows to. 

Such gateway, its very existence, implies that there is both a SIP Server and and XMPP server for each domain. It is a cross protocol federation, the domain must exists otherwise there is nothing to gateway to and there is no reason to deploy it. 

Adrian

>  
> Regards,
>  
> Christer
>  
>  
>  
>  
>  
> Otherwise you will move all SIP functionality into the XMPP gateway and must document that whole SIP universe as running inside that gateway, which makes little sense.
>  
> Regards, 
> Adrian
>  
> On Aug 2, 2013, at 12:16 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
> 
> 
> Hi,
>  
> As I indicated at the STOX session (and was later repeated by Jonathan Lennox), the draft need to have a story on SIP forking, e.g. if what happens if the INVITE sent from the interworking node gets forked, and multiple early dialogs are created.
>  
> Regards,
>  
> Christer
>  
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