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

[Christer Holmberg <[e-mail]>]

Hi,

I am not sure I understand.

The server may implement SIP functions like forking, and fork the INVITE. BUT, the gateway will still get the 18x responses for the early dialogs - just like a SIP phone.

Regards,

Christer

Lähettäjä: Adrian Georgescu [mailto:[e-mail]]
Lähetetty: 4. elokuuta 2013 20:20
Vastaanottaja: Christer Holmberg
Kopio: [e-mail]
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking

The gateway lookups the SIP server for a given domain in the DNS same as any other device would do like a SIP phone configured with credentials under that domain. Is the job of that server to properly implement all SIP functions like forking proxy, handle nat traversal or registrar to name a few common features.

You ask what shall the world do if the SIP server of given domain does not implemented forking or other feature. My answer is the gateway must not care, is the problem of whomever manages that domain and its applications to handle such features.

Regards,
Adrian



On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:


Hi,

There for sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) in the network, but how can you guarantee that it will handle forking? A forking proxy will simply forward 18x responses forming the early dialogs towards the gateway, which then will have to handle them.

Regards,

Christer

Lähettäjä: Adrian Georgescu [mailto:[e-mail]]
Lähetetty: 2. elokuuta 2013 13:36
Vastaanottaja: Christer Holmberg
Kopio: [e-mail]<mailto:[e-mail]>
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking


On Aug 2, 2013, at 12:30 PM, Christer Holmberg <[e-mail]<mailto:[e-mail]>> 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 <[e-mail]<mailto:[e-mail]>> 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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