Re: [Sip] Delivering request-URI and parameters to UAS via proxy

"Hans Erik van Elburg" <hanserik.van.elburg@ericsson.com> Fri, 01 February 2008 09:52 UTC

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Thread-Topic: [Sip] Delivering request-URI and parameters to UAS via proxy
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From: "Hans Erik van Elburg" <hanserik.van.elburg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <youssef.chadli@orange-ftgroup.com>
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Subject: Re: [Sip] Delivering request-URI and parameters to UAS via proxy
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Paul Kyzivat wrote:
> From the point of view of the registrar this seems similar to IMS implicit registration, 
> except that the scale is much larger - there may be *very many* implicit registrations 
> generated by this one explicit registration.

For this reason the Wildcarded Public User Identity concept is introduced in IMS. This allows efficient implicit registration of ranges of numbers or sets of sip uri's.

Best Regards,
/Hans Erik

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Wednesday, January 30, 2008 4:12 PM
To: youssef.chadli@orange-ftgroup.com
Cc: sip@ietf.org; drage@alcatel-lucent.com
Subject: Re: [Sip] Delivering request-URI and parameters to UAS via proxy



youssef.chadli@orange-ftgroup.com wrote:
> I think there is a misunderstanding on these use cases:

Quite possible.

>  - The intermediate entity in the customer side does perform only one registration to the network.
>    The AoRs associated to the served terminals inside the client domain are registred automatically
>    by the home network, using implicit registration for example. In fact, it's not conceivable to
>    register individually all the users of a corporate network for example.

You seem to be describing one of the models covered by the SIPForum in their SIPConnect specification. Is that what you mean?

 From the point of view of the registrar this seems similar to IMS implicit registration, except that the scale is much larger - there may be *very many* implicit registrations generated by this one explicit registration.

> - The "intermediate entity" should be able to route incoming requests inside its domain based on
>   standard RFC3261 behaviour, that is, based on the Request-URI.

I find the case much less compelling than the IMS case.

IMO this is an abuse of REGISTER to a purpose for which it is not well suited.

Treating it as registration means that there is an *expectation* that the R-URI will be translated. That isn't really appropriate in this case. The registration doesn't specify the mapping anyway - it is configuration in the network of all the addresses that belong to the "pbx" that does that. This is more appropriately viewed as just a routing operation, and so pushing the address of the enterprise server as a Route header is by far the best choice. Its only because this has been done using REGISTER that brings up how to preserve the old address when it is translated.

	Thanks,
	Paul

> Best regards
> 
> Youssef
> 
> 
>   
>  
> 
>> -----Message d'origine-----
>> De : Paul Kyzivat [mailto:pkyzivat@cisco.com] Envoyé : lundi 28 
>> janvier 2008 21:46 À : CHADLI Youssef RD-CORE-ISS Cc : 
>> drage@alcatel-lucent.com; sip@ietf.org Objet : Re: [Sip] Delivering 
>> request-URI and parameters to UAS via proxy
>>
>> These cases can all be handled by having intermediate entity use a 
>> unique user part, together with its own domain name, to construct a 
>> contact address for each terminal UA that it is registering. Then 
>> when it receives an incoming request to one of these uris it can 
>> simply translate it, using the user part as a key to its own local 
>> mappings.
>>
>> We don't need anything new to solve these cases.
>>
>> 	Paul
>>
>> youssef.chadli@orange-ftgroup.com wrote:
>>>  
>>> There are other cases similar to what is described in
>> section  2.1 Unknown Aliases of J. Rosenberg draft that should be 
>> taken into account.
>>> Those cases are where the customer side is composed of
>> several SIP terminals that access the network through an intermediate 
>> entity which registers their associated aliases with its contact 
>> address. Thus, such client side is seen from the network as a single 
>> client having several aliases (aggregated endpoints). Moreover, such 
>> client side may be a mini private network composed of several 
>> entities. In these configurations, the intermediate entity is in 
>> charge of routeing incoming calls inside the client domain and may 
>> need to behave as a SIP proxy for incoming SIP messages.
>>> As examples of such configuration:
>>> - Corporate networks: in that case the PBX register all the
>> served individual user identities with its contact address and routes 
>> incoming calls toward the individual called users.
>>> - Home networks: the Home Gateway may need to register on
>> behalf of all the served terminals.  The Home Gateway is in charge of 
>> routeing incoming calls toward the individual called users.
>>> J. Rosenberg draft seems give a good solution to take into
>> account these configurations.
>>>    
>>> Best regards,
>>>
>>> Youssef
>>>
>>>
>>>> -----Message d'origine-----
>>>> De : DRAGE, Keith (Keith)
>> [mailto:drage@alcatel-lucent.com] Envoyé : 
>>>> lundi 14 janvier 2008 17:58 À : sip@ietf.org Objet : [Sip]
>> Delivering
>>>> request-URI and parameters to UAS via proxy
>>>>
>>>> (As WG chair)
>>>>
>>>> In fulfilment of our charter items of
>>>>
>>>> Dec 2007    Delivering request-URI and parameters to UAS 
>> via proxy to
>>>> WGLC  
>>>> Feb 2008    Delivering request-URI and parameters to UAS 
>> via proxy to
>>>> IESG (PS)
>>>>
>>>> We now have a couple of proposals on the table for solving the 
>>>> problem.
>>>>
>>>> The original draft from Jonathan and which led to the
>> creation of the
>>>> charter items by the WG is unfortunately expired, but is at:
>>>>
>>>> http://tools.ietf.org/id/draft-rosenberg-sip-ua-loose-route-01.txt
>>>>
>>>> The alternative document from Christer, etc is at:
>>>>
>>>> http://www.ietf.org/internet-drafts/draft-holmberg-sip-target-
>>>> uri-delive
>>>> ry-00.txt
>>>>
>>>> We obviously need to make a decision between the two approaches so 
>>>> please attempt to address the following specific points via the 
>>>> mailing
>>>> list:
>>>>
>>>> 1)	Problem cases: These are summarised in section 4.4 of
>>>> draft-rosenberg-sip-ua-loose-route-01 and from my read of
>> the other
>>>> draft, I don't believe that this draft adds any others.
>>>> If you believe there are other cases that should be covered by the 
>>>> solution, then please identify them. If there is support
>> on any new
>>>> problem cases, I would encourage the authors of both drafts to add 
>>>> text concerning these problem cases.
>>>>
>>>> 2)	Clarifications: If for any reason you don't understand either
>>>> draft, or believe that there are technical issues that are not 
>>>> represented in the current draft, please post your questions / 
>>>> comments to the list. I would encourage authors of both drafts to 
>>>> revise as frequently as appropriate to reflect the current
>> state of
>>>> discussion.
>>>>
>>>> 3)	Support for either position. If you wish to indicate support for
>>>> either position please do so, but please accompany this is
>> technical
>>>> reasoning as to why you have this position, as that will
>> help other
>>>> members of the WG form a position.
>>>>
>>>> I would encourage as much list discussion as possible before 
>>>> Philadelphia. I suspect we will need to have a discussion at the 
>>>> face-to-face meeting in Philadephia, but list discussion
>> is essential
>>>> prior to that. If we can solve this positions on list,
>> then well and
>>>> good, and even better (that is how we are meant to make decisions).
>>>>
>>>> Regards
>>>>
>>>>
>>>> Keith
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use 
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>> sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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