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Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)

2010-07-01 07:54:15
In the PSTN,  we use the UUI field to transmit information between the
Intelligent Network (IN) system and call center agents for the
directory enquires service. Everybody in Germany who wants to ask for
the phone number of another person dials DT's directory enquires
service and is connected to a call center agent who tells him the
number he wants to know. Additionaly, only if the caller is a DT
customer, the call center agent offers to connect him to this number,
so the caller does not need to dial. For everybode else, he does not.
So the call center agent needs to get the information whether or not
the caller is a DT customer. This is done by routing every directory
enquires call to the IN system first. The IN system checks the caller
number and inserts the information about whether or not the caller is
a DT-customer in the UUI field which is transmited via INAP, ISUP and
DSS1 to the call center agent's end device.The call center agent gets
a display about this. During the PSTN-migration to SIP, we will have
cases where the call center and the IN system are connected to
different networks, one to PSTN and the other to SIP.  Also, we may
have applications as above on pure SIP application servers.

Can you shed some light on *how* this is used, given the lack of any
standards on the content/formatting of this information?

The application is DT-specific and needs a DT specification to be
supported by the IN system and the call center, but the container to
transport the information must be supported by both ends and by the
network nodes.

Or is this only used between a caller and a callee that have somehow
obtained contextual information that they both support this feature *and* a
particular encoding?

At least within the DT network, UUI is used between application
servers or application servers and "special" end devices as call
centers. As far as I know, UUI is currently not part of the DT normal
telephony package. Many years ago, it was, but people misused it.

We plan to use the "Johnston uui draft" for the scenario described
above and we see the need for the proposed WG.

Thanks a lot
Laura


Laura








2010/6/30 Paul Kyzivat <pkyzivat(_at_)cisco(_dot_)com>:
James,

Can you shed some light on *how* this is used, given the lack of any
standards on the content/formatting of this information?

Do you use content=isdn-uui and some particular Q.931 protocol discriminator
for which there are formatting standards?

Or is this only used between a caller and a callee that have somehow
obtained contextual information that they both support this feature *and* a
particular encoding?

       Thanks,
       Paul

James Rafferty wrote:

Hi,
My company has had the experience of deploying the pre-standard version of
this PSTN to SIP UUI mechanism during the past 2 years.
As noted in the draft charter, UUI information is widely used on the PSTN
for applications such as offering input data into call centers and then
preserving that data as calls get transferred.
Since many contact centers are now built using SIP, but still have PSTN
subscribers (via SS7 ISUP or ISDN PRI), there is a need to be able to
interwork the user to user information from the PSTN side into SIP.  In our
experience, the "Johnston uui draft" has accomplished this and we have
several customers that find this interworking to be valuable.  We also noted
improvements from early drafts into the later ones in areas such as making
better use of the ITU-T protocol discriminator, thus enabling better
interoperability from the PSTN side into SIP.
The major deficiency of the current draft is its non-standard status.
Functionally, we and our customers find this mechanism to be very useful and
I'd very much like the IETF to charter the a UUI work group to standardize
such a mechanism.
James  -----Original Message-----
From: dispatch-bounces(_at_)ietf(_dot_)org 
[mailto:dispatch-bounces(_at_)ietf(_dot_)org] On
Behalf Of Gonzalo Camarillo
Sent: Wednesday, June 30, 2010 2:51 AM
To: Paul Kyzivat
Cc: dispatch(_at_)ietf(_dot_)org; DRAGE, Keith (Keith); 
ietf(_at_)ietf(_dot_)org
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP
(cuss)

Hi,

please keep both the IETF and the DISPATCH mailing lists in the
recipients list in this discussion.

Cheers,

Gonzalo


On 29/06/2010 8:23 PM, Paul Kyzivat wrote:

DRAGE, Keith (Keith) wrote:

The UUI information is NOT ISUP. It is a transparent envelope to the
entire ISDN, so it is not part of an ISDN protocol and therefore not part 
of
an ISUP protocol. When carried by ISUP the envelope is bundled up in 
another
envelope with other information that ISUP carries transparently.

So I believe, and have said repeatedly in the past, that references to
SIP-T are irrelevant in this case.

The problem we do have though is that are legacy usages of this
information. In particular applications in PBXs carry it between themselves
in ISDN call establishment. The information itself is not standardised. If
you want to migrate a PBX from DSS1 to SIP, then you have to take this
information into account. The desire is not for a WG group to standardise
this existing usage (which in my view would be a complete non-starter), but
to allow the transfer of the existing information when migrated to a SIP
environment. The information transferred does not form part of SIP, and
should not be standardised as part of SIP.

How many different mechanisms do we have to construct for the purpose of
tunneling stuff over SIP?

Its especially bad if the stuff is neither standardized nor negotiated.
It then just provides more opportunity for non-interoperability.

It had been my impression that this content was standardized by ITU.
If nobody can bother to standardize it, then it hardly seems worth
working on.

       Thanks,
       Paul

regards

Keith



-----Original Message-----
From: dispatch-bounces(_at_)ietf(_dot_)org
[mailto:dispatch-bounces(_at_)ietf(_dot_)org] On Behalf Of
bruno(_dot_)chatras(_at_)orange-ftgroup(_dot_)com
Sent: Tuesday, June 29, 2010 1:00 PM
To: Gonzalo(_dot_)Camarillo(_at_)ericsson(_dot_)com; 
dispatch(_at_)ietf(_dot_)org
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI
for SIP (cuss)

Hum, I'm a bit surprised by the comment about SIP-T. RFC3372
does state that SIP-T does not come into play when there is
no PSTN involved.

At the end of clause 2 we can read the following: "4.  IP
origination - IP termination: This is a case for pure SIP.
SIP-T (either encapsulation or translation of ISUP) does not
come into play as there is no PSTN interworking."

Besides, SIP-T would require encapsulating a full ISUP
message (e.g. IAM) while we are interested in just one
parameter of the message in the context of CUSS. This would I
believe be a more severe drawback if SIP-T were used for this purpose.

Bruno


-----Message d'origine-----
De : dispatch-bounces(_at_)ietf(_dot_)org
[mailto:dispatch-bounces(_at_)ietf(_dot_)org] De la part de Gonzalo 
Camarillo
Envoyé : mardi 29 juin 2010 13:03 À : DISPATCH Objet :

[dispatch] Fwd:

Re: WG Review: Call Control UUI for SIP (cuss)

Hi,

FYI: note the discussion below on the IETF general list.

Cheers,

Gonzalo

-------- Original Message --------
Subject: Re: WG Review: Call Control UUI for SIP (cuss)
Date: Mon, 28 Jun 2010 20:24:23 +0200
From: Cullen Jennings <fluffy(_at_)cisco(_dot_)com>
To: iesg(_at_)ietf(_dot_)org <iesg(_at_)ietf(_dot_)org>
CC: IETF Discussion Mailing List <ietf(_at_)ietf(_dot_)org>


As far as I can tell, the WG says they wants to transfer some
information to achieve cross vendor interoperability.
However, what I believe the charter is actually going to do

is exactly

the opposite of that. When you get your head around what

this charter

is proposing, it is going to form a more or less opaque

container for

transporting proprietary information in a SIP header. It's hard to
imagine how this will help interoperability.

If we wanted to have interoperability, the charter would say what
information needs to be transferred and have the WG define a way to
get it between systems in an operable way.
What the charter for this WG actually says they are going to do is
make a special container for transfer proprietary information.
There's not even willing to say what that proprietary

information is

used for other than things ISDN UUI which is a  non

interoperable and

fairly proprietary field in ISDN.
Furthermore they have asserted that  existing containers

such as SIP-T

or SIP bodies can't be used for reasons that are hard to

describe. (I

reject the idea that because the call might not involved

the PSTN, it

can't use SIP-T).

I think the folks that want to do this should get a much clear
explanation of how this results in interoperability and why exist
approach such as SIP-T will not work before this is chartered.

I do think there is a need to standardize some important

call control

information used in call centers and related places.
However, the "we need a standard container to exchange secret
information WG" is a bad idea and violates the sprit of the

SIP change

process not to mention the mission of the IETF.

In summary, I'm in favor of figuring out what the problems

are people

hope to solve with this WG and figuring out a way to write
interoperable standards to achieve that. However, I think

this charter

should be rejected by the IESG and sent back to the drawing

board. The

RAI area has things of higher priority items to work on than a SIP
header for transfer proprietary data.



On Jun 22, 2010, at 10:00 , IESG Secretary wrote:

A new IETF working group has been proposed in the Real-time
Applications and Infrastructure Area.  The IESG has not

made any determination as yet.

The following draft charter was submitted, and is provided for
informational purposes only. Please send your comments to

the IESG

mailing list (iesg(_at_)ietf(_dot_)org) by Tuesday, June 29, 2010.

Call Control UUI for SIP  (cuss)
--------------------------------------------------
Current Status: Proposed Working Group

Last modified: 2010-06-21

Chair(s):
 TBD

Real-time Applications and Infrastructure Area Director(s):
 Gonzalo Camarillo <Gonzalo(_dot_)Camarillo(_at_)ericsson(_dot_)com>

Robert Sparks

<rjsparks(_at_)nostrum(_dot_)com>

Real-time Applications and Infrastructure Area Advisor:
 Gonzalo Camarillo <Gonzalo(_dot_)Camarillo(_at_)ericsson(_dot_)com>

Mailing Lists: TBD

Description of Working Group:

The Call Control UUI for SIP (CUSS) working group is chartered to
define a Session Initiation Protocol (SIP) mechanism for

transporting

call-control related user-to-user information (UUI) between User
Agents.

The mechanism developed in this working group is

applicable 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. The behavior of SIP entities that support it is not

significantly

 changed (as discussed in Section 4 of RFC 5727).
3. Generally only the User Agent Client (UAC) and User

Agent Server

 (UAS) are interested in the information.
4. The information is expected to survive retargeting,

redirection,

 and transfers.
5. SIP elements may need to apply policy about passing

and screening

 the information.
6. Multi-vendor interoperability is important.

This mechanism is not applicable in the following situations:

1. The behavior of SIP entities that support it is significantly
 changed (as discussed in Section 4 of RFC 5727).
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 in
 consuming (beyond applying policy) the information.
4. There are specific privacy issues involved with the information
 being transported (e.g., geolocation or emergency-related
 information).

User data of the mechanism will be clearly marked with the
application, encoding, semantics, and content type,

allowing policy to

be applied by UAs.  The working group will define the

information that

each application must specify to utilize the mechanism.

This type of

application-specific information will be specified in

standards-track

documents.

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 to

the PSTN-SIP

gateways.  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 since 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.

Even though interworking with the PSTN is an important

requirement,

call control UUI can be exchanged between native SIP

clients that do

not have any ISUP support. Therefore, existing SIP-T
encapsulation-based approaches defined in RFC3372 do not meet the
requirements to transport this type of information.

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
====================

Sep 10 - Problem statement and requirements document to IESG
(Informational)
Mar 11 - SIP call control UUI specification to IESG (PS)
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Cullen Jennings
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