Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP

"Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com> Wed, 04 November 2009 16:07 UTC

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Thread-Topic: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Cc: Leon Portman <Leon.Portman@nice.com>, dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for SIP
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Yeah this is a good case and for it we need re-Invite to carry it.

If it is session related data, then for transfers should it be carried
in REFER also, esp. for REFER with Replaces?

Sanjay

-----Original Message-----
From: Paul Kyzivat (pkyzivat) 
Sent: Wednesday, November 04, 2009 7:31 PM
To: Sanjay Sinha (sanjsinh)
Cc: Leon Portman; Alan Johnston; dispatch@ietf.org
Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP

I always use 3pcc as a test case, because it presents problems almost 
every time.

Of course the problem with 3pcc is that one "side" may remain the same, 
while the other "side" is replaced. The replacement typically has no 
knowledge of the history of what went on before it joined. So there 
usually needs to be some mechanism for it to be clued in about that.

I don't know if the same is true here or not. It depends on whether the 
UUI information is viewed as transient, or as state of the session.

	Thanks,
	Paul

Sanjay Sinha (sanjsinh) wrote:
> Do you mean same IE that was present in initial INVITE needs to be
> presented in Re-Invite as well? 
> 
> If the information does not change, then I do not see any value in
> carrying the same payload in re-Invite. But if there is a new IE, then
> yeah it can be presented in re-Invite.
> I am not sure if there can be a new UU IE after the call is setup and
> before call termination.
> 
> Sanjay
> 
> -----Original Message-----
> From: Leon Portman [mailto:Leon.Portman@nice.com] 
> Sent: Wednesday, November 04, 2009 3:02 PM
> To: Sanjay Sinha (sanjsinh); Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
> 
> Actually I was thinking that if UUI was presented during the original
> INVITE, if it is going to be required to be presented during
RE_INVITES
> as well, especially if it represents some client state information
> 
> Leon
> 
> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] 
> Sent: Wednesday, November 04, 2009 11:18 AM
> To: Leon Portman; Alan Johnston; dispatch@ietf.org
> Subject: RE: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
> 
> I think UUI is carried during session setup and teardown, and hence
> re-Invite scenario may not be applicable here.
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Leon Portman
> Sent: Wednesday, November 04, 2009 1:29 PM
> To: Alan Johnston; dispatch@ietf.org
> Subject: Re: [dispatch] Updated CCUS Charter Text - Call Control UUI
for
> SIP
> 
> Only one comment, does it also supposed to support re-invite (for SRTP
> re-keying or media updates) scenarios as well?
> 
> Thanks
> 
> Leon
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Alan Johnston
> Sent: Tuesday, November 03, 2009 11:09 PM
> To: dispatch@ietf.org
> Subject: [dispatch] Updated CCUS Charter Text - Call Control UUI for
SIP
> 
> All,
> 
> Here is some updated charter text for CCUS.  Comments are most
welcome.
> 
> Thanks,
> Alan
> 
> - - - - -
> 
> The Call Control UUI for SIP  (CCUS) working group is chartered to 
> define a SIP mechanism for transporting call control related 
> user-to-user (UUI) information between User Agents.
> 
> The mechanism developed in this working group is generally appropriate

> in the following situations:
> 1. The information is generated and consumed by an application using
SIP
> 
> during session setup but the application is not necessarily even SIP
> aware.
> 2. SIP behavior is unchanged.
> 3. Generally only the UAC and UAS are interested in the information.
> 4. The information is expected to survive retargeting, redirection,
and 
> retargeting.
> 5. SIP elements may need to apply policy about passing and screening
the
> 
> information.
> 6. Multi-vendor interoperability is important.
> 
> This mechanism is not appropriate in the following situations:
> 1. SIP behavior is changed.
> 2. The information is generated and consumed at the SIP layer by SIP 
> elements.
> 3. SIP elements besides the UAC and UAS might be interested and
consume 
> the information.
> 4. There are specific privacy or cross-RAI issues involved with the 
> information being transported.
> 
> User data of the mechanism will be clearly marked with the
application, 
> encoding, semantics, and contents of the data, allowing policy to be 
> applied by UAs.  The working group will define the information which 
> each application must specify in a standards-track document to utilize

> the mechanism.
> 
> One important application of this mechanism is interworking with the 
> ISDN User to User Information Service.  This application defined by 
> ITU-T Q.931, is extensively deployed in the PSTN today, supporting
such 
> applications as contact centers, call centers, and automatic call 
> distributors (ACDs).  A major barrier to the movement of these 
> applications to SIP is the lack of a standard mechanism to transport 
> this information in SIP.   For interworking with ISDN, minimal 
> information about the content of the UUI is available.  In this case 
> only, the content will just indicate ISDN UUI Service 1 interworking 
> rather than the actual content.
> 
> Call control UUI is user information conveyed between user agents
during
> 
> call control operations.  As a result, the information must be
conveyed 
> with the INVITE transaction, and must survive proxy retargeting, 
> redirection, and transfers.  The mechanism must utilize a minimum of
SIP
> 
> extensions as it will need to be supported by many simple SIP user 
> agents such as PSTN gateways.  The mechanism must interwork with the 
> existing ISDN service but should also be extensible for use by other 
> applications and non-ISDN protocols.
> 
> While PSTN UUI is carried in ISUP (ISDN User Part), SIP call control
UUI
> 
> can be generated by a number of protocols that are not ISUP, and as
such
> 
> it is architecturally unreasonable to interwork these protocols at the

> SIP / ISUP gateway. That is, existing SIP-T approaches defined in 
> RFC3372 do not meet the requirements.
> 
> Mechanisms based on the SIP INFO method, RFC2976, will not be
considered
> 
> by the working group since the UUI contents carry information that
must 
> be conveyed during session setup at the user agent - the information 
> must be conveyed with the INVITE transaction.  The information must be

> passed with the session setup request (INVITE), responses to that 
> INVITE, or session termination requests.  As a result, it is not 
> possible to use INFO in these cases.
> 
> The group will produce:
> 
> - A problem statement and requirements document for implementing a SIP

> call control UUI mechanism
> 
> - A specification of the SIP extension to best meet those
requirements.
> 
> Goals and Milestones
> ====================
> 
> Mar 10 - Problem statement and requirements document to IESG 
> (Informational)
> Aug 10 - SIP call control UUI specification to IESG (PS)
> 
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