
From gonzalo.camarillo@ericsson.com  Tue Jun  1 07:38:22 2010
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Date: Tue, 01 Jun 2010 17:38:00 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Dean Willis <dean.willis@softarmor.com>
References: <4BF3F281.8080102@ericsson.com> <4BF4E72B.7060600@ericsson.com>	 <AANLkTim7X6yFu-jZcOu9LFxgrGwY9mMAkJzhThMEijsp@mail.gmail.com> <1274663264.1419.6.camel@damnableubu>
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Subject: Re: [dispatch] Alert Info URNs - Acronym
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Hi Dean,

yes, this is a good one ;o)

Cheers,

Gonzalo

On 24/05/2010 4:07 AM, Dean Willis wrote:
> On Thu, 2010-05-20 at 10:17 -0500, Mary Barnes wrote:
>> What about SIPALE (SIP ALErting)?
>>
> 
> This suggests a variant, also drink related:
> 
> SALUD (Sip ALerting for User Devices)
> 
> "Â¡salud!" is a proper phrase for a very short toast, meaning "To your
> health!". Gonzalo should recognize it....
> 
> Besides, we're supposed to be multilingual these days, right?
> 
> --
> Dean
> 


From fluffy@cisco.com  Tue Jun  1 09:55:30 2010
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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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I've been talking to a lot of people about the VIPR drafts  - here is a =
first cut of a proposal for a WG that could do this. I'm sure the =
charter proposal needs a bunch of work but I wanted to get the =
discussion rolling on the list.=20

Thanks, Cullen=20

(PS - this is sent in my individual contributor role. Take all my posts =
about VIPR to be in my individual role not my co-chair role)

-------------------------------------------------------

ViPR Charter Proposal (Version 0)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications that =
more than a billion people use on a daily basis. They are phone numbers =
and DNS rooted address such as web servers and email addresses. The =
federation design of SIP is primarily designed for email style addresses =
yet a large percentage of SIP deployments primarily use phone numbers =
for identifying users. The goal of this working group is to allows =
people to use SIP to federate over the the internet while still using =
phone numbers to identify the person they wish to communicate with.=20

The VIPR WG will address this problem by developing a peer to peer based =
approach to finding SIP domains that claim to be responsible for a given =
phone number and the WG will design validation protocols to ensure a =
reasonable likelihood that a given domain actually is responsible for =
the phone number. One initial validation protocol will be based on a =
domain being able to prove it received a particular phone call over the =
PSTN based on both sides knowing the start and stop times of that call. =
Other validation schemes, such as examining fingerprints or watermarking =
of PSTN media, to show that a domain received a particular PSTN phone =
call may be considered by the working group. To help mitigate SPAM over =
SIP issues, the WG will define an token based authorization scheme so =
that domain using SIP to federate can choose to check that incoming SIP =
calls are from a domain that successfully validated a phone number.=20

The problem statement and some possible starting points for solutions =
are further desired in the following internet drafts which shall form =
the bases of the WG documents:
draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam

The working group will carefully coordinate with the security area, O&M =
area, as well as the appropriate RAI WG including sipcore and p2psip.=20



From mperumal@cisco.com  Tue Jun  1 11:12:26 2010
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Thread-Topic: [dispatch] Charter Proposal: Verification Involving PSTNReachability (VIPR)
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From: "Muthu ArulMozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "DISPATCH list" <dispatch@ietf.org>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTNReachability (VIPR)
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|The goal of this working group is to allows people to use SIP to
|federate over the the internet while still using phone numbers to=20
|identify the person they wish to communicate with.

Do we want to use phone numbers as the only way to identify persons or
allow people to use phone numbers as one of the ways to identify
persons?

If it is the later, should we reword "still using phone numbers to
identify" as "still allowing phone numbers to identify"?

thanks,
Muthu          =20

|-----Original Message-----
|From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
|Of Cullen Jennings (fluffy)
|Sent: Tuesday, June 01, 2010 10:25 PM
|To: DISPATCH list
|Subject: [dispatch] Charter Proposal: Verification Involving
|PSTNReachability (VIPR)
|
|
|I've been talking to a lot of people about the VIPR drafts  - here is a
|first cut of a proposal for a WG that could do this. I'm sure the
charter
|proposal needs a bunch of work but I wanted to get the discussion
rolling on
|the list.
|
|Thanks, Cullen
|
|(PS - this is sent in my individual contributor role. Take all my posts
|about VIPR to be in my individual role not my co-chair role)
|
|-------------------------------------------------------
|
|ViPR Charter Proposal (Version 0)
|
|WG Name:  Verification Involving PSTN Reachability (VIPR)
|
|There are two globally deployed address spaces for communications that
more
|than a billion people use on a daily basis. They are phone numbers and
DNS
|rooted address such as web servers and email addresses. The federation
|design of SIP is primarily designed for email style addresses yet a
large
|percentage of SIP deployments primarily use phone numbers for
identifying
|users. The goal of this working group is to allows people to use SIP to
|federate over the the internet while still using phone numbers to
identify
|the person they wish to communicate with.
|
|The VIPR WG will address this problem by developing a peer to peer
based
|approach to finding SIP domains that claim to be responsible for a
given
|phone number and the WG will design validation protocols to ensure a
|reasonable likelihood that a given domain actually is responsible for
the
|phone number. One initial validation protocol will be based on a domain
|being able to prove it received a particular phone call over the PSTN
based
|on both sides knowing the start and stop times of that call. Other
|validation schemes, such as examining fingerprints or watermarking of
PSTN
|media, to show that a domain received a particular PSTN phone call may
be
|considered by the working group. To help mitigate SPAM over SIP issues,
the
|WG will define an token based authorization scheme so that domain using
SIP
|to federate can choose to check that incoming SIP calls are from a
domain
|that successfully validated a phone number.
|
|The problem statement and some possible starting points for solutions
are
|further desired in the following internet drafts which shall form the
bases
|of the WG documents:
|draft-rosenberg-dispatch-vipr-overview
|draft-rosenberg-dispatch-vipr-reload-usage
|draft-rosenberg-dispatch-vipr-pvp
|draft-rosenberg-dispatch-vipr-sip-antispam
|
|The working group will carefully coordinate with the security area, O&M
|area, as well as the appropriate RAI WG including sipcore and p2psip.
|
|
|_______________________________________________
|dispatch mailing list
|dispatch@ietf.org
|https://www.ietf.org/mailman/listinfo/dispatch

From peter.musgrave@magorcorp.com  Tue Jun  1 11:29:12 2010
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Hi Cullen,

Does the charter need to say anything explicitly about not relying on
any of the work which uses ENUMs in DNS as a way of associating SIP
endpoints and IP addresses?

AFAIK ViPR is about *using* a SIP call to determine a path to the
endpoint (however that happens to work).

Does the charter need to stipulate that the mapping from a PSTN number
to a SIP device is a one-to-one mapping? Does there need to be text
about how multiple endpoints which are behind a border device (and all
appear as the same PSTN number) are to be handled?.

ViPR so far describes an "edge-device" approach. Is the charter
restricted to this approach?

Thanks,

Peter Musgrave



On Tue, Jun 1, 2010 at 12:55 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
> I've been talking to a lot of people about the VIPR drafts =A0- here is a=
 first cut of a proposal for a WG that could do this. I'm sure the charter =
proposal needs a bunch of work but I wanted to get the discussion rolling o=
n the list.
>
> Thanks, Cullen
>
> (PS - this is sent in my individual contributor role. Take all my posts a=
bout VIPR to be in my individual role not my co-chair role)
>
> -------------------------------------------------------
>
> ViPR Charter Proposal (Version 0)
>
> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>
> There are two globally deployed address spaces for communications that mo=
re than a billion people use on a daily basis. They are phone numbers and D=
NS rooted address such as web servers and email addresses. The federation d=
esign of SIP is primarily designed for email style addresses yet a large pe=
rcentage of SIP deployments primarily use phone numbers for identifying use=
rs. The goal of this working group is to allows people to use SIP to federa=
te over the the internet while still using phone numbers to identify the pe=
rson they wish to communicate with.
>
> The VIPR WG will address this problem by developing a peer to peer based =
approach to finding SIP domains that claim to be responsible for a given ph=
one number and the WG will design validation protocols to ensure a reasonab=
le likelihood that a given domain actually is responsible for the phone num=
ber. One initial validation protocol will be based on a domain being able t=
o prove it received a particular phone call over the PSTN based on both sid=
es knowing the start and stop times of that call. Other validation schemes,=
 such as examining fingerprints or watermarking of PSTN media, to show that=
 a domain received a particular PSTN phone call may be considered by the wo=
rking group. To help mitigate SPAM over SIP issues, the WG will define an t=
oken based authorization scheme so that domain using SIP to federate can ch=
oose to check that incoming SIP calls are from a domain that successfully v=
alidated a phone number.
>
> The problem statement and some possible starting points for solutions are=
 further desired in the following internet drafts which shall form the base=
s of the WG documents:
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
>
> The working group will carefully coordinate with the security area, O&M a=
rea, as well as the appropriate RAI WG including sipcore and p2psip.
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From pkyzivat@cisco.com  Tue Jun  1 12:09:11 2010
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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Peter Musgrave wrote:
> Hi Cullen,
> 
> Does the charter need to say anything explicitly about not relying on
> any of the work which uses ENUMs in DNS as a way of associating SIP
> endpoints and IP addresses?
> 
> AFAIK ViPR is about *using* a SIP call to determine a path to the
> endpoint (however that happens to work).

As I understand it, the new WG is about coming up with technique(s) that 
work and are deployable in the real world. While in principle public 
ENUM could be used for this, in practice it isn't deployed and so isn't 
really usable. The technique of using a pstn call to determine the path 
is one technique that is demonstrably deployable.

> Does the charter need to stipulate that the mapping from a PSTN number
> to a SIP device is a one-to-one mapping? Does there need to be text
> about how multiple endpoints which are behind a border device (and all
> appear as the same PSTN number) are to be handled?.

IIUC, its likely to be a server responsible for a phone number AOR that 
will participate in ViPR, rather than individual phone devices. 
Certainly there can be additional routing after a ViPR call reaches that 
server. Typically there will only be one of those, or a cooperative 
cluster of them sharing the same DNS name or ip address. That is really 
no different than normal sip routing to AORs.

Also, AFAIK it doesn't have to be 1:1.
Multiple claims can be made for ownership of a phone number. If calls to 
them can be verified, then they can all be valid candidates for calls to 
that number. (However, this will logistically be hard to accomplish.)

	Thanks,
	Paul

> ViPR so far describes an "edge-device" approach. Is the charter
> restricted to this approach?
> 
> Thanks,
> 
> Peter Musgrave
> 
> 
> 
> On Tue, Jun 1, 2010 at 12:55 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>> I've been talking to a lot of people about the VIPR drafts  - here is a first cut of a proposal for a WG that could do this. I'm sure the charter proposal needs a bunch of work but I wanted to get the discussion rolling on the list.
>>
>> Thanks, Cullen
>>
>> (PS - this is sent in my individual contributor role. Take all my posts about VIPR to be in my individual role not my co-chair role)
>>
>> -------------------------------------------------------
>>
>> ViPR Charter Proposal (Version 0)
>>
>> WG Name:  Verification Involving PSTN Reachability (VIPR)
>>
>> There are two globally deployed address spaces for communications that more than a billion people use on a daily basis. They are phone numbers and DNS rooted address such as web servers and email addresses. The federation design of SIP is primarily designed for email style addresses yet a large percentage of SIP deployments primarily use phone numbers for identifying users. The goal of this working group is to allows people to use SIP to federate over the the internet while still using phone numbers to identify the person they wish to communicate with.
>>
>> The VIPR WG will address this problem by developing a peer to peer based approach to finding SIP domains that claim to be responsible for a given phone number and the WG will design validation protocols to ensure a reasonable likelihood that a given domain actually is responsible for the phone number. One initial validation protocol will be based on a domain being able to prove it received a particular phone call over the PSTN based on both sides knowing the start and stop times of that call. Other validation schemes, such as examining fingerprints or watermarking of PSTN media, to show that a domain received a particular PSTN phone call may be considered by the working group. To help mitigate SPAM over SIP issues, the WG will define an token based authorization scheme so that domain using SIP to federate can choose to check that incoming SIP calls are from a domain that successfully validated a phone number.
>>
>> The problem statement and some possible starting points for solutions are further desired in the following internet drafts which shall form the bases of the WG documents:
>> draft-rosenberg-dispatch-vipr-overview
>> draft-rosenberg-dispatch-vipr-reload-usage
>> draft-rosenberg-dispatch-vipr-pvp
>> draft-rosenberg-dispatch-vipr-sip-antispam
>>
>> The working group will carefully coordinate with the security area, O&M area, as well as the appropriate RAI WG including sipcore and p2psip.
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From laura.liess.dt@googlemail.com  Tue Jun  1 22:54:35 2010
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This sounds good.
Laura


2010/6/1 Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>:
> Hi Dean,
>
> yes, this is a good one ;o)
>
> Cheers,
>
> Gonzalo
>
> On 24/05/2010 4:07 AM, Dean Willis wrote:
>> On Thu, 2010-05-20 at 10:17 -0500, Mary Barnes wrote:
>>> What about SIPALE (SIP ALErting)?
>>>
>>
>> This suggests a variant, also drink related:
>>
>> SALUD (Sip ALerting for User Devices)
>>
>> "=A1salud!" is a proper phrase for a very short toast, meaning "To your
>> health!". Gonzalo should recognize it....
>>
>> Besides, we're supposed to be multilingual these days, right?
>>
>> --
>> Dean
>>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From deepanshu@huawei.com  Wed Jun  2 00:50:15 2010
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Date: Wed, 02 Jun 2010 15:49:33 +0800
From: Deepanshu Gautam <deepanshu@huawei.com>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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Should there be a set of deliverables  what this WG is intending for? Or the
draft mentioned are actually the deliverables for this WG.

>The goal of this working group is to allows people to use SIP to federate
over the the internet while still using phone numbers to identify the person
they wish to communicate with.

I would suggest to emphasize(for clarity) that the federation being refered
here is the *across-domain* federation (although, some wise guys can figure
it out with *over the interent* :-))

Does charter have to say that, some extensions(headers, option-tag, ...) to
SIP are invisioned e.g for transfering a piece-of-information meant to
authorize particular domain to contact a particuar phone number?

Regards
Deepanshu


-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Cullen Jennings
Sent: 2010$BG/(B6$B7n(B2$BF|(B 0:55
To: DISPATCH list
Subject: [dispatch] Charter Proposal: Verification Involving PSTN
Reachability (VIPR)


I've been talking to a lot of people about the VIPR drafts  - here is a
first cut of a proposal for a WG that could do this. I'm sure the charter
proposal needs a bunch of work but I wanted to get the discussion rolling on
the list. 

Thanks, Cullen 

(PS - this is sent in my individual contributor role. Take all my posts
about VIPR to be in my individual role not my co-chair role)

-------------------------------------------------------

ViPR Charter Proposal (Version 0)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications that more
than a billion people use on a daily basis. They are phone numbers and DNS
rooted address such as web servers and email addresses. The federation
design of SIP is primarily designed for email style addresses yet a large
percentage of SIP deployments primarily use phone numbers for identifying
users. The goal of this working group is to allows people to use SIP to
federate over the the internet while still using phone numbers to identify
the person they wish to communicate with. 

The VIPR WG will address this problem by developing a peer to peer based
approach to finding SIP domains that claim to be responsible for a given
phone number and the WG will design validation protocols to ensure a
reasonable likelihood that a given domain actually is responsible for the
phone number. One initial validation protocol will be based on a domain
being able to prove it received a particular phone call over the PSTN based
on both sides knowing the start and stop times of that call. Other
validation schemes, such as examining fingerprints or watermarking of PSTN
media, to show that a domain received a particular PSTN phone call may be
considered by the working group. To help mitigate SPAM over SIP issues, the
WG will define an token based authorization scheme so that domain using SIP
to federate can choose to check that incoming SIP calls are from a domain
that successfully validated a phone number. 

The problem statement and some possible starting points for solutions are
further desired in the following internet drafts which shall form the bases
of the WG documents:
draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam

The working group will carefully coordinate with the security area, O&M
area, as well as the appropriate RAI WG including sipcore and p2psip. 


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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Folks,

Here is a version of the work description that addresses the comments
made on the first draft.=20

The changes include:=20

1.       Charter section, qualification that we are working on SIP-based
systems

2.       Charter section, minor word smithing

3.       Charter section, reference to RAI WG with relevant work

4.       Scope section, clarification of interoperability requirements

5.       Scope section, replacement of second paragraph regarding
treatment of non-video non-audio media types

6.       Goals and Milestones - addition of a Problem Statement draft

=20

=20

Comments welcome

=20

Allyn

=20

Multi-streams for Telepresence Description of Work

=20

=20

=20

Background

One branch of video conferencing has evolved that is focused on
immersive "being there" experience.  Referred to in various ways such as
virtual conferencing, telepresence or media spaces, early systems were
mainly research projects or business systems with limited deployments.
In recent years telepresence systems have seen considerable market
success.  Following the model of early systems, the first wave of
commercial systems have been typically located in specially designed
single-purpose rooms with multiple relatively large displays permitting
life size image reproduction, multiple cameras, encoders, decoders and
microphones.  These systems have several important characteristics that
are different from more traditional video conferencing systems. =20

=20

The first difference concerns controlling the visual viewpoint in order
to improve participant nonverbal communication. These systems preserve
essential group meeting characteristics such as eye contact, group
gestures, seating order and spatial audio by carefully orchestrating the
miking and camera angles at each of the sites . This is distinct from
the more traditional approach where the geometric relationship between
media streams is not used to preserve inter-stream communication aspects
such as eye contact and group dynamics. =20

=20

A second difference is manipulation of the environment to improve
immersion.  With telepresence systems, cinematographic aspects of the
local environment reproduction are carefully planned including color,
table shape, seating and lighting so that when combined with large high
quality displays, a strong sense of a "trompe l'oeil" or "being there"
immersive experience is created.  Typical video conference systems do
not include these considerations.

=20

As telepresence video systems have become successful in the market,
manufacturers have started exploring delivery of the nonverbal
communication and immersive values of telepresence via smaller, less
expensive and more flexible video conferencing systems for a variety of
venues, such as individual offices, homes and kiosks. These are also
telepresence systems, since the audio and video quality is high enough
to allow clear image reproduction for nonverbal communication, they are
able to send and receive multiple media streams, and large coordinated
multi image displays are available for immersive installations.   As the
industry develops, the line between telepresence and video conferencing
may become blurred as nonverbal communication and immersive
installations become broadly available.

=20

Problem

Although telepresence systems are based on open standards such as RTP,
SIP, H.264, H.323 suite, they cannot easily interoperate with each other
without operator assistance and expensive additional equipment that
translates from one vendor to another. It would be like having to make
sure all parties are on the same equipment (and network) when making a
telephone call.  A major factor in the inability of Telepresence systems
to work with each other is that there is no standard description of the
multiple streams that comprise the media flows.=20

=20

For example, in a multiple screen conference, the video and audio
streams sent from remote participants must be understood by receivers so
that they can be presented in a coherent and life-like manner. This
includes the ability to present the remote participants at their true
size for their apparent distance, while maintaining correct eye contact,
gesticular cues, and simultaneously providing a spatial audio sound
stage consistent with the video presentation.  The receiving device that
decides how to display the incoming information needs to understand a
number of variables such as the spatial position of the speaker, the
field of view of the cameras, the camera zoom, which media stream is
related to each of the displays, etc.=20

=20

Charter

The Telepresence Multi-Streams work item in DISPATCH is chartered to
define and specify for SIP-based systems the content of media
multi-stream messages and the way these will be transported.=20

=20

This work will provide a standard for the exchange of media semantic
information that will foster interoperable end stations and conference
bridges. It will specify  variables that describe the semantics of the
media streams and the recommended behavior to achieve interoperability.


=20

This requires considering current widely deployed use cases, such as
single and multiple screens, multi-point, Scalable Video Coding (SVC),
as well as cases that are expected to be implemented using the protocol
framework produced by this work item.  The methodology for describing
the variables must allow extensibility of the variables, since
telepresence is still a young technology and may have use cases that are
not currently considered.

=20

The work item will identify use cases, define requirements, and define a
method for describing and transporting information about multiple media
streams, including a specification of variables required to support the
use cases. This work item will consider the reuse of existing IETF
protocols and produce an architecture/protocol framework document
describing the protocols required to be implemented to support this
functionality.  The document will identify any enhancements required to
existing protocols as well as describing new protocol(s) for
interoperable multi-streams negotiation that may be required.

=20

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and
FECframe.

=20

Scope

The scope includes both systems that provide a fully immersive
experience, and systems that interwork with them and therefore need to
understand the same multiple stream semantics.

=20

=20

The focus of this work is on audio and video multiple streams.  Other
media types may be considered, however development of methodologies for
them is not within the scope of this work.

=20

Interoperation with standards compliant systems is required, such as
SIP-based video conferencing systems.  However, backwards compatibility
with existing non-standards compliant telepresence systems is not
required.

=20

=20

=20

The group will produce

- Requirements and use cases

=20

- Architectural Framework describing the protocols required to be
implemented to support this functionality and identifying existing
protocol enhancements and new protocol functionality required

=20

- Specification of a new protocol to support telepresence multi-streams
[if required]

=20

Goals and Milestones

Nov 2010=20

=20

Use Cases and Requirements to IESG as Informational RFC=20

Nov 2010

March 2011=20

Problem Statement to IESG as Informational RFC

Architecture to IESG as Informational RFC=20

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011=20

Submit protocol draft to IESG as Proposed Standard RFC=20

=20

=20

=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal>Folks,<o:p></o:p></p>

<p class=3DMsoNormal>Here is a version of the work description that =
addresses the
comments made on the first draft. <o:p></o:p></p>

<p class=3DMsoNormal>The changes include: <o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Charter section, qualification that we are =
working on
SIP-based systems<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Charter section, minor word =
smithing<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Charter section, reference to RAI WG with =
relevant work<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Scope section, clarification of interoperability
requirements<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>5.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Scope section, replacement of second paragraph
regarding treatment of non-video non-audio media types<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>6.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Goals and Milestones &#8211; addition of a =
Problem
Statement draft<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Comments welcome<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Allyn<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'>Multi-streams for =
Telepresence
Description of Work<o:p></o:p></span></b></p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Background<o:p></o:p></span></=
b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>One branch of
video conferencing has evolved that is focused on immersive &#8220;being
there&#8221; experience.&nbsp; Referred to in various ways such as =
virtual
conferencing, telepresence or media spaces, early systems were mainly =
research
projects or business systems with limited deployments.&nbsp; In recent =
years
telepresence systems have seen considerable market success.&nbsp; =
Following the
model of early systems, the first wave of commercial systems have been
typically located in specially designed single-purpose rooms with =
multiple
relatively large displays permitting life size image reproduction, =
multiple
cameras, encoders, decoders and microphones.&nbsp; These systems have =
several
important characteristics that are different from more traditional video
conferencing systems.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The first
difference concerns controlling the visual viewpoint in order to improve
participant nonverbal communication. These systems preserve essential =
group
meeting characteristics such as eye contact, group gestures, seating =
order and
spatial audio by carefully orchestrating the miking and camera angles at =
each
of the sites . This is distinct from the more traditional approach where =
the geometric
relationship between media streams is not used to preserve inter-stream
communication aspects such as eye contact and group dynamics.&nbsp; =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>A =
second
difference is manipulation of the environment to improve =
immersion.&nbsp; With
telepresence systems, cinematographic aspects of the local environment
reproduction are carefully planned including color, table shape, seating =
and
lighting so that when combined with large high quality displays, a =
strong sense
of a &quot;trompe l'oeil&quot; or &quot;being there&quot; immersive =
experience
is created.&nbsp; Typical video conference systems do not include these
considerations.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>As
telepresence video systems have become successful in the market, =
manufacturers
have started exploring delivery of the nonverbal communication and =
immersive
values of telepresence via smaller, less expensive and more flexible =
video
conferencing systems for a variety of venues, such as individual =
offices, homes
and kiosks. These are also&nbsp; telepresence systems, since the audio =
and
video quality is high enough to allow clear image reproduction for =
nonverbal
communication, they are able to send and receive multiple media streams, =
and
large coordinated multi image displays are available for immersive
installations.&nbsp;&nbsp; As the industry develops, the line between =
telepresence
and video conferencing may become blurred as nonverbal communication and
immersive installations become broadly available.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Problem<o:p></o:p></span></b><=
/p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Although
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323
suite, they cannot easily interoperate with each other without operator
assistance and expensive additional equipment that translates from one =
vendor
to another. It would be like having to make sure all parties are on the =
same
equipment (and network) when making a telephone call. &nbsp;A major =
factor in
the inability of Telepresence systems to work with each other is that =
there is
no standard description of the multiple streams that comprise the media =
flows. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>For example,
in a multiple screen conference, the video and audio streams sent from =
remote
participants must be understood by receivers so that they can be =
presented in a
coherent and life-like manner. This includes the ability to present the =
remote
participants at their true size for their apparent distance, while =
maintaining
correct eye contact, gesticular cues, and simultaneously providing a =
spatial
audio sound stage consistent with the video presentation.&nbsp; The =
receiving
device that decides how to display the incoming information needs to =
understand
a number of variables such as the spatial position of the speaker, the =
field of
view of the cameras, the camera zoom, which media stream is related to =
each of
the displays, etc. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Charter<o:p></o:p></span></b><=
/p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The
Telepresence Multi-Streams work item in DISPATCH is chartered to define =
and
specify for SIP-based systems the content of media multi-stream messages =
and
the way these will be transported. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>This
work will provide a standard for the exchange of media semantic =
information
that will foster interoperable end stations and conference bridges. =
</span><span
style=3D'font-family:"Arial","sans-serif"'>It will specify&nbsp; =
variables that
describe the semantics of the media streams and the recommended behavior =
to
achieve interoperability.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>This
requires considering current widely deployed use cases, such as single =
and
multiple screens, multi-point, Scalable Video Coding (SVC), as well as =
cases
that are expected to be implemented using the protocol framework =
produced by
this work item.&nbsp; The methodology for describing the variables must =
allow
extensibility of the variables, since telepresence is still a young =
technology
and may have use cases that are not currently =
considered.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>The
work item will identify use cases, define requirements, and define a =
method for
describing and transporting information about multiple media streams, =
including
a specification of variables required to support the use cases. This =
work item
will consider the reuse of existing IETF protocols and produce an
architecture/protocol framework document describing the protocols =
required to
be implemented to support this functionality.&nbsp; The document will =
identify
any enhancements required to existing protocols as well as describing =
new
protocol(s) for interoperable multi-streams negotiation that may be =
required.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>Relevant
work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><b><span =
style=3D'font-family:
"Arial","sans-serif"'>Scope<o:p></o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The scope
includes both systems that provide a fully immersive experience, and =
systems
that interwork with them and therefore need to understand the same =
multiple
stream semantics.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'>The
focus of this work is on audio and video multiple streams.&nbsp; Other =
media
types may be considered, however development of methodologies for them =
is not
within the scope of this work.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:12.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>Interoperation
with standards compliant systems is required, such as SIP-based video
conferencing systems. &nbsp;However, backwards compatibility with =
existing
non-standards compliant telepresence systems is not =
required.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><b><span =
style=3D'font-family:
"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:#0070C0'><o:p>&nbsp;</o:p=
></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>The group
will produce<o:p></o:p></span></b></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Requirements and use cases<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Architectural Framework describing the protocols required to be =
implemented to
support this functionality and identifying existing protocol =
enhancements and
new protocol functionality required<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Specification of a new protocol to support telepresence multi-streams =
[if
required]<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Goals and
Milestones<o:p></o:p></span></b></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2010 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Use Cases
  and Requirements to IESG as Informational RFC </span><span =
style=3D'font-size:
  12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2010</span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Problem
  Statement to IESG as Informational RFC</span><span =
style=3D'font-size:12.0pt;
  font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Architecture
  to IESG as Informational RFC </span><span =
style=3D'font-size:12.0pt;font-family:
  "Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011</span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Revise
  Charter [IF new protocol is not required]</span><span =
style=3D'font-size:12.0pt;
  font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2011 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Submit
  protocol draft to IESG as Proposed Standard RFC </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
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Hi Gao,

=20

I had a chance to talk with Mary about your idea to split up the work, =
and she confirmed that it is good practice in general to do so, but we =
thought it is as yet too early to split up the multi-stream work.  As we =
progress with use cases, problem statement and requirements, we will be =
able to see better if the topic should be split up.

=20

Best regards,

Allyn

=20

From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
Sent: Tuesday, May 18, 2010 10:33 PM
To: Allyn Romanow (allyn)
Cc: IETF DISPATCH list
Subject: Re: [dispatch] Multi-streams for telepresence description of =
work

=20


Hi,=20

I have two level of comments,=20

1. Feeling for the topic=20
Technically, I am quit glad to see such discussion about telepresence, =
which is trailblazer for VR(Virtual Reality) production.=20

2. About the charter WG=20
One revolutionary application, such telepresence, would cover basic =
protocol level, application control level, and interworking issue.=20

If the work is just focused on pure interworking issue, then the topics =
can be divided into smaller one, which might be in other basic =
protocols' WG's scope. But if the application itself is new, and has =
some new characteristic, more work should be proposed.=20

If the work is also focused on telepresence application, which is a =
conferencing system, the xcon WG might be proper one for application =
level definition.=20

If the charter also want to talk the esential aspect of telepresence(or =
more abstractly, about VR), such as how to using SDP, or relative =
protocol to describe the streams and it's collaboration, it would be =
quite new topic. One new WG might be better.=20


I find the charter seems covers all the three thing. So, I suggest you =
classify such different level requirement. Then, we may find which part =
is proper for which WG, and which part should be covered by new WG.=20

Thanks,=20

Gao=20

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"Allyn Romanow (allyn)" <allyn@cisco.com>=20
=B7=A2=BC=FE=C8=CB:  dispatch-bounces@ietf.org=20

2010-05-19 02:30=20

=CA=D5=BC=FE=C8=CB

"IETF DISPATCH list" <dispatch@ietf.org>=20

=B3=AD=CB=CD

=09
=D6=F7=CC=E2

[dispatch] Multi-streams for telepresence description of work

=20

	=09




Hi Folks,=20
 =20
Many of you are aware of telepresence systems (video conferencing on =
steroids..:), fewer probably are aware of the fact that the current =
systems are not interoperable with each other. After talking with Mary =
and Cullen, we have a rough draft of a charter to work on this problem =
in DISPATCH,  for discussion on the mailing list.=20
 =20
We don=A1=AFt yet know whether a working group would need to be spun out =
or not =A8C that will be clarified as we go.=20
 =20
We think there is enough background info in the work description to =
start a discussion, and plan to send out use cases for discussion =
shortly.=20
 =20
Who is the we? Several makers and providers of telepresence systems have =
gotten together to define the main problem in interoperability =A8C the =
handling of multi-streams.=20
 =20
Thanks for your thoughts-=20
Allyn=20
 =20
 =20


Multi-streams for Telepresence Description of Work=20
 =20
=20


 =20
Background=20
One branch of video conferencing has evolved that is focused on =
immersive =A1=B0being there=A1=B1 experience.  Referred to in various =
ways such as virtual conferencing, telepresence or media spaces, early =
systems were mainly research projects or business systems with limited =
deployments.  In recent years telepresence systems have seen =
considerable market success.  Following the model of early systems, the =
first wave of commercial systems have been typically located in =
specially designed single-purpose rooms with multiple relatively large =
displays permitting life size image reproduction, multiple cameras, =
encoders, decoders and microphones.  These systems have several =
important characteristics that are different from more traditional video =
conferencing systems.  =20
 =20
The first difference concerns controlling the visual viewpoint in order =
to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group =
gestures, seating order and spatial audio by carefully orchestrating the =
miking and camera angles at each of the sites . This is distinct from =
the more traditional approach where the geometric relationship between =
media streams is not used to preserve inter-stream communication aspects =
such as eye contact and group dynamics.  =20
 =20
A second difference is manipulation of the environment to improve =
immersion.  With telepresence systems, cinematographic aspects of the =
local environment reproduction are carefully planned including color, =
table shape, seating and lighting so that when combined with large high =
quality displays, a strong sense of a "trompe l'oeil" or "being there" =
immersive experience is created.  Typical video conference systems do =
not include these considerations.=20
 =20
As telepresence video systems have become successful in the market, =
manufacturers have started exploring delivery of the nonverbal =
communication and immersive values of telepresence via smaller, less =
expensive and more flexible video conferencing systems for a variety of =
venues, such as individual offices, homes and kiosks. These are also  =
telepresence systems, since the audio and video quality is high enough =
to allow clear image reproduction for nonverbal communication, they are =
able to send and receive multiple media streams, and large coordinated =
multi image displays are available for immersive installations.   As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.=20
 =20
Problem=20
Although telepresence systems are based on open standards such as RTP, =
SIP, H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call.  A major factor in the inability of Telepresence systems =
to work with each other is that there is no standard description of the =
multiple streams that comprise the media flows.=20
 =20
For example, in a multiple screen conference, the video and audio =
streams sent from remote participants must be understood by receivers so =
that they can be presented in a coherent and life-like manner. This =
includes the ability to present the remote participants at their true =
size for their apparent distance, while maintaining correct eye contact, =
gesticular cues, and simultaneously providing a spatial audio sound =
stage consistent with the video presentation.  The receiving device that =
decides how to display the incoming information needs to understand a =
number of variables such as the spatial position of the speaker, the =
field of view of the cameras, the camera zoom, which media stream is =
related to each of the displays, etc.=20
 =20
Charter=20
The Telepresence Multi-Streams work item in DISPATCH is chartered to =
define and specify the content of media multi-stream messages and the =
way these will be transported.=20
This work is aimed at providing a standard for the exchange of media =
semantics information that will foster interoperable end stations and =
conference bridges. This work item will specify the variables that =
describe the semantics of the media streams and the recommended behavior =
to achieve interoperability.  =20
 =20
This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  The methodology for describing =
the variables must allow extensibility of the variables, since =
telepresence is still a young technology and may have use cases that are =
not currently considered.=20
 =20
The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media =
streams, including a specification of variables required to support the =
use cases. This work item will consider the reuse of existing IETF =
protocols and produce an architecture/protocol framework document =
describing the protocols required to be implemented to support this =
functionality.  The document will identify any enhancements required to =
existing protocols as well as describing new protocol(s) for =
interoperable multi-streams negotiation that may be required.=20
 =20
Scope=20
The scope includes both systems that provide a fully immersive =
experience, and systems that interwork with them and therefore need to =
understand the same multiple stream semantics.=20
 =20
The focus of this work is on audio and video multiple streams, however =
other media types may be considered.  Definition of new application =
protocols is not within the scope of this work=20
 =20
[Is there anything else we need to explicitly say is out of scope?]=20
 =20
The group will produce=20
- Requirements and use cases=20
 =20
- Architectural Framework describing the protocols required to be =
implemented to support this functionality and identifying existing =
protocol enhancements and new protocol functionality required=20
 =20
- Specification of a new protocol to support telepresence multi-streams =
[if required]=20
 =20
Goals and Milestones=20

Nov 2010=20

Use Cases and Requirements to IESG as Informational RFC=20

March 2011=20

Architecture to IESG as Informational RFC=20

March 2011=20

Revise Charter [IF new protocol is not required]=20

Nov 2011=20

Submit protocol draft to IESG as Proposed Standard RFC=20


 =20
 =20
 =20
 _______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch



=20
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This email and any files transmitted with it are confidential and =
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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Gao,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I had a chance to talk with Mary about your idea to split =
up the
work, and she confirmed that it is good practice in general to do so, =
but we
thought it is as yet too early to split up the multi-stream work.&nbsp; =
As we
progress with use cases, problem statement and requirements, we will be =
able to
see better if the topic should be split up.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Best regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Allyn<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn] <br>
<b>Sent:</b> Tuesday, May 18, 2010 10:33 PM<br>
<b>To:</b> Allyn Romanow (allyn)<br>
<b>Cc:</b> IETF DISPATCH list<br>
<b>Subject:</b> Re: [dispatch] Multi-streams for telepresence =
description of
work<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Hi,</span> =
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I have =
two
level of comments,</span> <br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>1. =
Feeling for
the topic</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Technically, =
I
am quit glad to see such discussion about telepresence, which is =
trailblazer
for VR(Virtual Reality) production. </span><br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>2. =
About the
charter WG</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>One
revolutionary application, such telepresence, would cover basic protocol =
level,
application control level, and interworking issue. </span><br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>If the =
work is
just focused on pure interworking issue, then the topics can be divided =
into
smaller one, which might be in other basic protocols' WG's scope. But if =
the
application itself is new, and has some new characteristic, more work =
should be
proposed. </span><br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>If the =
work is
also focused on telepresence application, which is a conferencing =
system, the
xcon WG might be proper one for application level definition.</span> =
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>If the =
charter
also want to talk the esential aspect of telepresence(or more =
abstractly, about
VR), such as how to using SDP, or relative protocol to describe the =
streams and
it's collaboration, it would be quite new topic. One new WG might be =
better.</span>
<br>
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I find =
the
charter seems covers all the three thing. So, I suggest you classify =
such
different level requirement. Then, we may find which part is proper for =
which
WG, and which part should be covered by new WG.</span> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,</span=
> <br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Gao</span> =
<br>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<br>
Zip &nbsp; &nbsp;: 210012<br>
Tel &nbsp; &nbsp;: 87211<br>
Tel2 &nbsp; :(+86)-025-52877211<br>
e_mail : gao.yang2@zte.com.cn<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span> <br>
<br>
<o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0 width=3D"100%"
 style=3D'width:100.0%'>
 <tr>
  <td width=3D"26%" valign=3Dtop style=3D'width:26.0%;padding:.75pt =
.75pt .75pt .75pt'>
  <p class=3DMsoNormal><b><span =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif"'>&quot;Allyn
  Romanow (allyn)&quot; &lt;allyn@cisco.com&gt;</span></b><span
  style=3D'font-size:7.5pt;font-family:"Arial","sans-serif"'> =
</span><br>
  <span lang=3DZH-CN =
style=3D'font-size:7.5pt'>=B7=A2=BC=FE=C8=CB</span><span =
style=3D'font-size:
  7.5pt;font-family:"Arial","sans-serif"'>: =
&nbsp;dispatch-bounces@ietf.org</span>
  <o:p></o:p></p>
  <p><span =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif"'>2010-05-19
  02:30</span> <o:p></o:p></p>
  </td>
  <td width=3D"73%" valign=3Dtop style=3D'width:73.0%;padding:.75pt =
.75pt .75pt .75pt'>
  <table class=3DMsoNormalTable border=3D0 cellpadding=3D0 =
width=3D"100%"
   style=3D'width:100.0%'>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><span =
lang=3DZH-CN
    style=3D'font-size:7.5pt'>=CA=D5=BC=FE=C8=CB</span><o:p></o:p></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><span =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif"'>&quot;IETF
    DISPATCH list&quot; &lt;dispatch@ietf.org&gt;</span> <o:p></o:p></p>
    </td>
   </tr>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><span =
lang=3DZH-CN
    style=3D'font-size:7.5pt'>=B3=AD=CB=CD</span><o:p></o:p></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'></td>
   </tr>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><span =
lang=3DZH-CN
    style=3D'font-size:7.5pt'>=D6=F7=CC=E2</span><o:p></o:p></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><span =
style=3D'font-size:7.5pt;font-family:"Arial","sans-serif"'>[dispatch]
    Multi-streams for telepresence description of =
work</span><o:p></o:p></p>
    </td>
   </tr>
  </table>
  <p class=3DMsoNormal><o:p>&nbsp;</o:p></p>
  <table class=3DMsoNormalTable border=3D0 cellpadding=3D0>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'></td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'></td>
   </tr>
  </table>
  </td>
 </tr>
</table>

<p class=3DMsoNormal><br>
<br>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Hi =
Folks,</span>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Many =
of you are
aware of telepresence systems (video conferencing on steroids..:), fewer
probably are aware of the fact that the current systems are not =
interoperable
with each other. After talking with Mary and Cullen, we have a rough =
draft of a
charter to work on this problem in DISPATCH, &nbsp;for discussion on the
mailing list.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We =
don=A1=AFt yet
know whether a working group would need to be spun out or not =A8C that =
will be
clarified as we go.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We =
think there
is enough background info in the work description to start a discussion, =
and
plan to send out use cases for discussion shortly.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Who is =
the we?
Several makers and providers of telepresence systems have gotten =
together to
define the main problem in interoperability =A8C the handling of =
multi-streams.</span>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks =
for your
thoughts-</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Allyn</span> =
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Multi-streams=

for Telepresence Description of Work</span></b> <br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
</b>
<br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
</b><o:p></o:p></p>

<p class=3DMsoNormal><br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
</b>
<br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Background</s=
pan></b>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>One =
branch of
video conferencing has evolved that is focused on immersive =A1=B0being =
there=A1=B1
experience. &nbsp;Referred to in various ways such as virtual =
conferencing,
telepresence or media spaces, early systems were mainly research =
projects or
business systems with limited deployments. &nbsp;In recent years =
telepresence
systems have seen considerable market success. &nbsp;Following the model =
of
early systems, the first wave of commercial systems have been typically =
located
in specially designed single-purpose rooms with multiple relatively =
large
displays permitting life size image reproduction, multiple cameras, =
encoders,
decoders and microphones. &nbsp;These systems have several important =
characteristics
that are different from more traditional video conferencing systems. =
&nbsp;</span>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
first
difference concerns controlling the visual viewpoint in order to improve
participant nonverbal communication. These systems preserve essential =
group
meeting characteristics such as eye contact, group gestures, seating =
order and
spatial audio by carefully orchestrating the miking and camera angles at =
each
of the sites . This is distinct from the more traditional approach where =
the
geometric relationship between media streams is not used to preserve
inter-stream communication aspects such as eye contact and group =
dynamics.
&nbsp;</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A =
second
difference is manipulation of the environment to improve immersion. =
&nbsp;With
telepresence systems, cinematographic aspects of the local environment
reproduction are carefully planned including color, table shape, seating =
and
lighting so that when combined with large high quality displays, a =
strong sense
of a &quot;trompe l'oeil&quot; or &quot;being there&quot; immersive =
experience
is created. &nbsp;Typical video conference systems do not include these
considerations.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>As =
telepresence
video systems have become successful in the market, manufacturers have =
started
exploring delivery of the nonverbal communication and immersive values =
of
telepresence via smaller, less expensive and more flexible video =
conferencing
systems for a variety of venues, such as individual offices, homes and =
kiosks.
These are also &nbsp;telepresence systems, since the audio and video =
quality is
high enough to allow clear image reproduction for nonverbal =
communication, they
are able to send and receive multiple media streams, and large =
coordinated
multi image displays are available for immersive installations. &nbsp; =
As the
industry develops, the line between telepresence and video conferencing =
may
become blurred as nonverbal communication and immersive installations =
become
broadly available.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Problem</span=
></b>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Although
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323
suite, they cannot easily interoperate with each other without operator
assistance and expensive additional equipment that translates from one =
vendor
to another. It would be like having to make sure all parties are on the =
same
equipment (and network) when making a telephone call. &nbsp;A major =
factor in
the inability of Telepresence systems to work with each other is that =
there is
no standard description of the multiple streams that comprise the media =
flows. </span><br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>For =
example, in
a multiple screen conference, the video and audio streams sent from =
remote
participants must be understood by receivers so that they can be =
presented in a
coherent and life-like manner. This includes the ability to present the =
remote
participants at their true size for their apparent distance, while =
maintaining
correct eye contact, gesticular cues, and simultaneously providing a =
spatial
audio sound stage consistent with the video presentation. &nbsp;The =
receiving
device that decides how to display the incoming information needs to =
understand
a number of variables such as the spatial position of the speaker, the =
field of
view of the cameras, the camera zoom, which media stream is related to =
each of
the displays, etc. </span><br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Charter</span=
></b>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The
Telepresence Multi-Streams work item in DISPATCH is chartered to define =
and
specify the content of media multi-stream messages and the way these =
will be
transported. </span><br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>This =
work is
aimed at providing a standard for the exchange of media semantics =
information
that will foster interoperable end stations and conference bridges. This =
work
item will specify the variables that describe the semantics of the media
streams and the recommended behavior to achieve interoperability. =
&nbsp;</span>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>This =
requires
considering current widely deployed use cases, such as single and =
multiple
screens, multi-point, Scalable Video Coding (SVC), as well as cases that =
are
expected to be implemented using the protocol framework produced by this =
work
item. &nbsp;The methodology for describing the variables must allow
extensibility of the variables, since telepresence is still a young =
technology
and may have use cases that are not currently considered.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
work item
will identify use cases, define requirements, and define a method for
describing and transporting information about multiple media streams, =
including
a specification of variables required to support the use cases. This =
work item
will consider the reuse of existing IETF protocols and produce an
architecture/protocol framework document describing the protocols =
required to
be implemented to support this functionality. &nbsp;The document will =
identify
any enhancements required to existing protocols as well as describing =
new
protocol(s) for interoperable multi-streams negotiation that may be =
required.</span>
<br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Scope</span><=
/b>
<br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
scope
includes both systems that provide a fully immersive experience, and =
systems
that interwork with them and therefore need to understand the same =
multiple
stream semantics.</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
focus of
this work is on audio and video multiple streams, however other media =
types may
be considered. &nbsp;Definition of new application protocols is not =
within the
scope of this work</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>[Is =
there
anything else we need to explicitly say is out of scope?]</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:#0082BF'=
>&nbsp;</span>
<br>
<b><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
group
will produce</span></b> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>- =
Requirements
and use cases</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>- =
Architectural
Framework describing the protocols required to be implemented to support =
this
functionality and identifying existing protocol enhancements and new =
protocol
functionality required</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>- =
Specification
of a new protocol to support telepresence multi-streams [if =
required]</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<b><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Goals and
Milestones</span></b> <o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0 width=3D"100%"
 style=3D'width:100.0%'>
 <tr>
  <td width=3D"20%" style=3D'width:20.0%;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Nov
  2010 </span><o:p></o:p></p>
  </td>
  <td width=3D"79%" style=3D'width:79.0%;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Use
  Cases and Requirements to IESG as Informational RFC =
</span><o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>March
  2011 </span><o:p></o:p></p>
  </td>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Architecture
  to IESG as Informational RFC </span><o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>March
  2011</span> <o:p></o:p></p>
  </td>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Revise
  Charter [IF new protocol is not required]</span> <o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Nov
  2011 </span><o:p></o:p></p>
  </td>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Submit
  protocol draft to IESG as Proposed Standard RFC </span><o:p></o:p></p>
  </td>
 </tr>
</table>

<p style=3D'margin-bottom:12.0pt'><br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
 <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<tt><span
style=3D'font-size:10.0pt'>______________________________________________=
_</span></tt><span
style=3D'font-size:10.0pt'><br>
<tt>dispatch mailing list</tt><br>
<tt>dispatch@ietf.org</tt><br>
<tt>https://www.ietf.org/mailman/listinfo/dispatch</tt><br>
<br>
</span><o:p></o:p></p>

<pre><o:p>&nbsp;</o:p></pre><pre>----------------------------------------=
----------------<o:p></o:p></pre><pre>ZTE&nbsp;Information&nbsp;Security&=
nbsp;Notice:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&n=
bsp;mail&nbsp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's=
&nbsp;organization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;c=
onfidential.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligate=
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itted&nbsp;to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp=
;communication&nbsp;to&nbsp;others.<o:p></o:p></pre><pre>This&nbsp;email&=
nbsp;and&nbsp;any&nbsp;files&nbsp;transmitted&nbsp;with&nbsp;it&nbsp;are&=
nbsp;confidential&nbsp;and&nbsp;intended&nbsp;solely&nbsp;for&nbsp;the&nb=
sp;use&nbsp;of&nbsp;the&nbsp;individual&nbsp;or&nbsp;entity&nbsp;to&nbsp;=
whom&nbsp;they&nbsp;are&nbsp;addressed.&nbsp;If&nbsp;you&nbsp;have&nbsp;r=
eceived&nbsp;this&nbsp;email&nbsp;in&nbsp;error&nbsp;please&nbsp;notify&n=
bsp;the&nbsp;originator&nbsp;of&nbsp;the&nbsp;message.&nbsp;Any&nbsp;view=
s&nbsp;expressed&nbsp;in&nbsp;this&nbsp;message&nbsp;are&nbsp;those&nbsp;=
of&nbsp;the&nbsp;individual&nbsp;sender.<o:p></o:p></pre><pre>This&nbsp;m=
essage&nbsp;has&nbsp;been&nbsp;scanned&nbsp;for&nbsp;viruses&nbsp;and&nbs=
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From peter.musgrave@magorcorp.com  Wed Jun  2 09:40:19 2010
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To: "Allyn Romanow (allyn)" <allyn@cisco.com>
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Subject: Re: [dispatch] Updated charter/work description for telepresence multi-streams
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Hi,

I read this through. I think it describes the work well enough at this
point.

Is it worth narrowing the scope of how media will be transmitted? I assume
it will be RTP but this not explicitly stated.

Regards,

Peter Musgrave

On Wed, Jun 2, 2010 at 12:11 PM, Allyn Romanow (allyn) <allyn@cisco.com>wro=
te:

>  Folks,
>
> Here is a version of the work description that addresses the comments mad=
e
> on the first draft.
>
> The changes include:
>
> 1.       Charter section, qualification that we are working on SIP-based
> systems
>
> 2.       Charter section, minor word smithing
>
> 3.       Charter section, reference to RAI WG with relevant work
>
> 4.       Scope section, clarification of interoperability requirements
>
> 5.       Scope section, replacement of second paragraph regarding
> treatment of non-video non-audio media types
>
> 6.       Goals and Milestones =96 addition of a Problem Statement draft
>
>
>
>
>
> Comments welcome
>
>
>
> Allyn
>
>
>
> *Multi-streams for Telepresence Description of Work*
>
> * *
>
> * *
>
> * *
>
> *Background*
>
> One branch of video conferencing has evolved that is focused on immersive
> =93being there=94 experience.  Referred to in various ways such as virtua=
l
> conferencing, telepresence or media spaces, early systems were mainly
> research projects or business systems with limited deployments.  In recen=
t
> years telepresence systems have seen considerable market success.  Follow=
ing
> the model of early systems, the first wave of commercial systems have bee=
n
> typically located in specially designed single-purpose rooms with multipl=
e
> relatively large displays permitting life size image reproduction, multip=
le
> cameras, encoders, decoders and microphones.  These systems have several
> important characteristics that are different from more traditional video
> conferencing systems.
>
>
>
> The first difference concerns controlling the visual viewpoint in order t=
o
> improve participant nonverbal communication. These systems preserve
> essential group meeting characteristics such as eye contact, group gestur=
es,
> seating order and spatial audio by carefully orchestrating the miking and
> camera angles at each of the sites . This is distinct from the more
> traditional approach where the geometric relationship between media strea=
ms
> is not used to preserve inter-stream communication aspects such as eye
> contact and group dynamics.
>
>
>
> A second difference is manipulation of the environment to improve
> immersion.  With telepresence systems, cinematographic aspects of the loc=
al
> environment reproduction are carefully planned including color, table sha=
pe,
> seating and lighting so that when combined with large high quality displa=
ys,
> a strong sense of a "trompe l'oeil" or "being there" immersive experience=
 is
> created.  Typical video conference systems do not include these
> considerations.
>
>
>
> As telepresence video systems have become successful in the market,
> manufacturers have started exploring delivery of the nonverbal communicat=
ion
> and immersive values of telepresence via smaller, less expensive and more
> flexible video conferencing systems for a variety of venues, such as
> individual offices, homes and kiosks. These are also  telepresence system=
s,
> since the audio and video quality is high enough to allow clear image
> reproduction for nonverbal communication, they are able to send and recei=
ve
> multiple media streams, and large coordinated multi image displays are
> available for immersive installations.   As the industry develops, the li=
ne
> between telepresence and video conferencing may become blurred as nonverb=
al
> communication and immersive installations become broadly available.
>
>
>
> *Problem*
>
> Although telepresence systems are based on open standards such as RTP, SI=
P,
> H.264, H.323 suite, they cannot easily interoperate with each other witho=
ut
> operator assistance and expensive additional equipment that translates fr=
om
> one vendor to another. It would be like having to make sure all parties a=
re
> on the same equipment (and network) when making a telephone call.  A majo=
r
> factor in the inability of Telepresence systems to work with each other i=
s
> that there is no standard description of the multiple streams that compri=
se
> the media flows.
>
>
>
> For example, in a multiple screen conference, the video and audio streams
> sent from remote participants must be understood by receivers so that the=
y
> can be presented in a coherent and life-like manner. This includes the
> ability to present the remote participants at their true size for their
> apparent distance, while maintaining correct eye contact, gesticular cues=
,
> and simultaneously providing a spatial audio sound stage consistent with =
the
> video presentation.  The receiving device that decides how to display the
> incoming information needs to understand a number of variables such as th=
e
> spatial position of the speaker, the field of view of the cameras, the
> camera zoom, which media stream is related to each of the displays, etc.
>
>
>
> *Charter*
>
> The Telepresence Multi-Streams work item in DISPATCH is chartered to defi=
ne
> and specify for SIP-based systems the content of media multi-stream messa=
ges
> and the way these will be transported.
>
>
>
> This work will provide a standard for the exchange of media semantic
> information that will foster interoperable end stations and conference
> bridges. It will specify  variables that describe the semantics of the
> media streams and the recommended behavior to achieve interoperability.
>
>
>
> This requires considering current widely deployed use cases, such as sing=
le
> and multiple screens, multi-point, Scalable Video Coding (SVC), as well a=
s
> cases that are expected to be implemented using the protocol framework
> produced by this work item.  The methodology for describing the variables
> must allow extensibility of the variables, since telepresence is still a
> young technology and may have use cases that are not currently considered=
.
>
>
>
> The work item will identify use cases, define requirements, and define a
> method for describing and transporting information about multiple media
> streams, including a specification of variables required to support the u=
se
> cases. This work item will consider the reuse of existing IETF protocols =
and
> produce an architecture/protocol framework document describing the protoc=
ols
> required to be implemented to support this functionality.  The document w=
ill
> identify any enhancements required to existing protocols as well as
> describing new protocol(s) for interoperable multi-streams negotiation th=
at
> may be required.
>
>
>
> Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and
> FECframe.
>
>
>
> *Scope*
>
> The scope includes both systems that provide a fully immersive experience=
,
> and systems that interwork with them and therefore need to understand the
> same multiple stream semantics.
>
>
>
>
>
> The focus of this work is on audio and video multiple streams.  Other med=
ia
> types may be considered, however development of methodologies for them is
> not within the scope of this work.
>
>
>
> Interoperation with standards compliant systems is required, such as
> SIP-based video conferencing systems.  However, backwards compatibility w=
ith
> existing non-standards compliant telepresence systems is not required.
>
>
>
> * *
>
>
>
> *The group will produce*
>
> - Requirements and use cases
>
>
>
> - Architectural Framework describing the protocols required to be
> implemented to support this functionality and identifying existing protoc=
ol
> enhancements and new protocol functionality required
>
>
>
> - Specification of a new protocol to support telepresence multi-streams [=
if
> required]
>
>
>
> *Goals and Milestones*
>
> Nov 2010
>
>
>
> Use Cases and Requirements to IESG as Informational RFC
>
> Nov 2010
>
> March 2011
>
> Problem Statement to IESG as Informational RFC
>
> Architecture to IESG as Informational RFC
>
> March 2011
>
> Revise Charter [IF new protocol is not required]
>
> Nov 2011
>
> Submit protocol draft to IESG as Proposed Standard RFC
>
>
>
>
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>

--000e0cd379aa93e98304880ebd77
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

Hi,=A0<div><br></div><div>I read this through. I think it describes the wor=
k well enough at this point.=A0</div><div><br></div><div>Is it worth narrow=
ing the scope of how media will be transmitted? I assume it will be RTP but=
 this not explicitly stated.</div>
<div><br></div><div>Regards,</div><div><br></div><div>Peter Musgrave<br><br=
><div class=3D"gmail_quote">On Wed, Jun 2, 2010 at 12:11 PM, Allyn Romanow =
(allyn) <span dir=3D"ltr">&lt;<a href=3D"mailto:allyn@cisco.com">allyn@cisc=
o.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">








<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">

<div>

<p class=3D"MsoNormal">Folks,</p>

<p class=3D"MsoNormal">Here is a version of the work description that addre=
sses the
comments made on the first draft. </p>

<p class=3D"MsoNormal">The changes include: </p>

<p><span>1.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Charter section, qualification that we are working on
SIP-based systems</p>

<p><span>2.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Charter section, minor word smithing</p>

<p><span>3.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Charter section, reference to RAI WG with relevant work</p>

<p><span>4.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Scope section, clarification of interoperability
requirements</p>

<p><span>5.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Scope section, replacement of second paragraph
regarding treatment of non-video non-audio media types</p>

<p><span>6.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=A0=A0=A0=
=A0=A0=A0
</span></span>Goals and Milestones =96 addition of a Problem
Statement draft</p>

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal">Comments welcome</p>

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal">Allyn</p>

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><b><spa=
n>Multi-streams for Telepresence
Description of Work</span></b></p>

<p class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><b><spa=
n>=A0</span></b></p>

<p class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><b><spa=
n>=A0</span></b></p>

<p class=3D"MsoNormal"><b><span>=A0</span></b></p>

<p class=3D"MsoNormal"><b><span>Background</span></b></p>

<p class=3D"MsoNormal"><span>One branch of
video conferencing has evolved that is focused on immersive =93being
there=94 experience.=A0 Referred to in various ways such as virtual
conferencing, telepresence or media spaces, early systems were mainly resea=
rch
projects or business systems with limited deployments.=A0 In recent years
telepresence systems have seen considerable market success.=A0 Following th=
e
model of early systems, the first wave of commercial systems have been
typically located in specially designed single-purpose rooms with multiple
relatively large displays permitting life size image reproduction, multiple
cameras, encoders, decoders and microphones.=A0 These systems have several
important characteristics that are different from more traditional video
conferencing systems.=A0 </span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><span>The first
difference concerns controlling the visual viewpoint in order to improve
participant nonverbal communication. These systems preserve essential group
meeting characteristics such as eye contact, group gestures, seating order =
and
spatial audio by carefully orchestrating the miking and camera angles at ea=
ch
of the sites . This is distinct from the more traditional approach where th=
e geometric
relationship between media streams is not used to preserve inter-stream
communication aspects such as eye contact and group dynamics.=A0 </span></p=
>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><span>A second
difference is manipulation of the environment to improve immersion.=A0 With
telepresence systems, cinematographic aspects of the local environment
reproduction are carefully planned including color, table shape, seating an=
d
lighting so that when combined with large high quality displays, a strong s=
ense
of a &quot;trompe l&#39;oeil&quot; or &quot;being there&quot; immersive exp=
erience
is created.=A0 Typical video conference systems do not include these
considerations.</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><span>As
telepresence video systems have become successful in the market, manufactur=
ers
have started exploring delivery of the nonverbal communication and immersiv=
e
values of telepresence via smaller, less expensive and more flexible video
conferencing systems for a variety of venues, such as individual offices, h=
omes
and kiosks. These are also=A0 telepresence systems, since the audio and
video quality is high enough to allow clear image reproduction for nonverba=
l
communication, they are able to send and receive multiple media streams, an=
d
large coordinated multi image displays are available for immersive
installations.=A0=A0 As the industry develops, the line between telepresenc=
e
and video conferencing may become blurred as nonverbal communication and
immersive installations become broadly available.</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><b><span>Problem</span></b></p>

<p class=3D"MsoNormal"><span>Although
telepresence systems are based on open standards such as RTP, SIP, H.264, H=
.323
suite, they cannot easily interoperate with each other without operator
assistance and expensive additional equipment that translates from one vend=
or
to another. It would be like having to make sure all parties are on the sam=
e
equipment (and network) when making a telephone call. =A0A major factor in
the inability of Telepresence systems to work with each other is that there=
 is
no standard description of the multiple streams that comprise the media flo=
ws. </span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><span>For example,
in a multiple screen conference, the video and audio streams sent from remo=
te
participants must be understood by receivers so that they can be presented =
in a
coherent and life-like manner. This includes the ability to present the rem=
ote
participants at their true size for their apparent distance, while maintain=
ing
correct eye contact, gesticular cues, and simultaneously providing a spatia=
l
audio sound stage consistent with the video presentation.=A0 The receiving
device that decides how to display the incoming information needs to unders=
tand
a number of variables such as the spatial position of the speaker, the fiel=
d of
view of the cameras, the camera zoom, which media stream is related to each=
 of
the displays, etc. </span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><b><span>Charter</span></b></p>

<p class=3D"MsoNormal"><span>The
Telepresence Multi-Streams work item in DISPATCH is chartered to define and
specify for SIP-based systems the content of media multi-stream messages an=
d
the way these will be transported. </span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>This
work will provide a standard for the exchange of media semantic information
that will foster interoperable end stations and conference bridges. </span>=
<span>It will specify=A0 variables that
describe the semantics of the media streams and the recommended behavior to
achieve interoperability.=A0 </span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>This
requires considering current widely deployed use cases, such as single and
multiple screens, multi-point, Scalable Video Coding (SVC), as well as case=
s
that are expected to be implemented using the protocol framework produced b=
y
this work item.=A0 The methodology for describing the variables must allow
extensibility of the variables, since telepresence is still a young technol=
ogy
and may have use cases that are not currently considered.</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>The
work item will identify use cases, define requirements, and define a method=
 for
describing and transporting information about multiple media streams, inclu=
ding
a specification of variables required to support the use cases. This work i=
tem
will consider the reuse of existing IETF protocols and produce an
architecture/protocol framework document describing the protocols required =
to
be implemented to support this functionality.=A0 The document will identify
any enhancements required to existing protocols as well as describing new
protocol(s) for interoperable multi-streams negotiation that may be require=
d.</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>Relevant
work to be drawn upon has been done in XCON, MMUSIC, AVT, and FECframe.</sp=
an></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><b><span>Scope</span><=
/b></p>

<p class=3D"MsoNormal"><span>The scope
includes both systems that provide a fully immersive experience, and system=
s
that interwork with them and therefore need to understand the same multiple
stream semantics.</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p><span style=3D"font-size:12.0pt">The
focus of this work is on audio and video multiple streams.=A0 Other media
types may be considered, however development of methodologies for them is n=
ot
within the scope of this work.</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span style=3D"font-si=
ze:12.0pt">=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>Interoperation
with standards compliant systems is required, such as SIP-based video
conferencing systems. =A0However, backwards compatibility with existing
non-standards compliant telepresence systems is not required.</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><b><span>=A0</span></b=
></p>

<p class=3D"MsoNormal"><span style=3D"color:#0070C0">=A0</span></p>

<p class=3D"MsoNormal"><b><span>The group
will produce</span></b></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>-
Requirements and use cases</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>-
Architectural Framework describing the protocols required to be implemented=
 to
support this functionality and identifying existing protocol enhancements a=
nd
new protocol functionality required</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>=A0</span></p>

<p class=3D"MsoNormal" style=3D"text-autospace:none"><span>-
Specification of a new protocol to support telepresence multi-streams [if
required]</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><b><span>Goals and
Milestones</span></b></p>

<table border=3D"0" cellpadding=3D"0">
 <tbody><tr>
  <td width=3D"96" style=3D"width:1.0in;padding:.75pt .75pt .75pt .75pt">
  <p class=3D"MsoNormal"><span>Nov 2010 </span><span style=3D"font-size:12.=
0pt"></span></p>
  </td>
  <td width=3D"381" style=3D"width:285.75pt;padding:.75pt .75pt .75pt .75pt=
">
  <p class=3D"MsoNormal"><span style=3D"font-size:12.0pt">=A0</span></p>
  <p class=3D"MsoNormal"><span>Use Cases
  and Requirements to IESG as Informational RFC </span><span style=3D"font-=
size:12.0pt"></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D"96" style=3D"width:1.0in;padding:.75pt .75pt .75pt .75pt">
  <p class=3D"MsoNormal"><span>Nov 2010</span><span style=3D"font-size:12.0=
pt"></span></p>
  <p class=3D"MsoNormal"><span>March 2011 </span><span style=3D"font-size:1=
2.0pt"></span></p>
  </td>
  <td width=3D"381" style=3D"width:285.75pt;padding:.75pt .75pt .75pt .75pt=
">
  <p class=3D"MsoNormal"><span>Problem
  Statement to IESG as Informational RFC</span><span style=3D"font-size:12.=
0pt"></span></p>
  <p class=3D"MsoNormal"><span>Architecture
  to IESG as Informational RFC </span><span style=3D"font-size:12.0pt"></sp=
an></p>
  </td>
 </tr>
 <tr>
  <td width=3D"96" style=3D"width:1.0in;padding:.75pt .75pt .75pt .75pt">
  <p class=3D"MsoNormal"><span>March 2011</span><span style=3D"font-size:12=
.0pt"></span></p>
  </td>
  <td width=3D"381" style=3D"width:285.75pt;padding:.75pt .75pt .75pt .75pt=
">
  <p class=3D"MsoNormal"><span>Revise
  Charter [IF new protocol is not required]</span><span style=3D"font-size:=
12.0pt"></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D"96" style=3D"width:1.0in;padding:.75pt .75pt .75pt .75pt">
  <p class=3D"MsoNormal"><span>Nov 2011 </span><span style=3D"font-size:12.=
0pt"></span></p>
  </td>
  <td width=3D"381" style=3D"width:285.75pt;padding:.75pt .75pt .75pt .75pt=
">
  <p class=3D"MsoNormal"><span>Submit
  protocol draft to IESG as Proposed Standard RFC </span><span style=3D"fon=
t-size:12.0pt"></span></p>
  </td>
 </tr>
</tbody></table>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal"><span>=A0</span></p>

<p class=3D"MsoNormal">=A0</p>

</div>

</div>


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<br></blockquote></div><br></div>

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From allyn@cisco.com  Wed Jun  2 09:45:19 2010
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Hi Peter,

It's certainly possible imo to explicitly state RTP. We could put it on
the list for changes, if there is no objection.

=20

Allyn

=20

From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]=20
Sent: Wednesday, June 02, 2010 9:40 AM
To: Allyn Romanow (allyn)
Cc: dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings
(fluffy)
Subject: Re: [dispatch] Updated charter/work description for
telepresence multi-streams

=20

Hi,=20

=20

I read this through. I think it describes the work well enough at this
point.=20

=20

Is it worth narrowing the scope of how media will be transmitted? I
assume it will be RTP but this not explicitly stated.

=20

Regards,

=20

Peter Musgrave

On Wed, Jun 2, 2010 at 12:11 PM, Allyn Romanow (allyn) <allyn@cisco.com>
wrote:

Folks,

Here is a version of the work description that addresses the comments
made on the first draft.=20

The changes include:=20

1.       Charter section, qualification that we are working on SIP-based
systems

2.       Charter section, minor word smithing

3.       Charter section, reference to RAI WG with relevant work

4.       Scope section, clarification of interoperability requirements

5.       Scope section, replacement of second paragraph regarding
treatment of non-video non-audio media types

6.       Goals and Milestones - addition of a Problem Statement draft

=20

=20

Comments welcome

=20

Allyn

=20

Multi-streams for Telepresence Description of Work

=20

=20

=20

Background

One branch of video conferencing has evolved that is focused on
immersive "being there" experience.  Referred to in various ways such as
virtual conferencing, telepresence or media spaces, early systems were
mainly research projects or business systems with limited deployments.
In recent years telepresence systems have seen considerable market
success.  Following the model of early systems, the first wave of
commercial systems have been typically located in specially designed
single-purpose rooms with multiple relatively large displays permitting
life size image reproduction, multiple cameras, encoders, decoders and
microphones.  These systems have several important characteristics that
are different from more traditional video conferencing systems. =20

=20

The first difference concerns controlling the visual viewpoint in order
to improve participant nonverbal communication. These systems preserve
essential group meeting characteristics such as eye contact, group
gestures, seating order and spatial audio by carefully orchestrating the
miking and camera angles at each of the sites . This is distinct from
the more traditional approach where the geometric relationship between
media streams is not used to preserve inter-stream communication aspects
such as eye contact and group dynamics. =20

=20

A second difference is manipulation of the environment to improve
immersion.  With telepresence systems, cinematographic aspects of the
local environment reproduction are carefully planned including color,
table shape, seating and lighting so that when combined with large high
quality displays, a strong sense of a "trompe l'oeil" or "being there"
immersive experience is created.  Typical video conference systems do
not include these considerations.

=20

As telepresence video systems have become successful in the market,
manufacturers have started exploring delivery of the nonverbal
communication and immersive values of telepresence via smaller, less
expensive and more flexible video conferencing systems for a variety of
venues, such as individual offices, homes and kiosks. These are also
telepresence systems, since the audio and video quality is high enough
to allow clear image reproduction for nonverbal communication, they are
able to send and receive multiple media streams, and large coordinated
multi image displays are available for immersive installations.   As the
industry develops, the line between telepresence and video conferencing
may become blurred as nonverbal communication and immersive
installations become broadly available.

=20

Problem

Although telepresence systems are based on open standards such as RTP,
SIP, H.264, H.323 suite, they cannot easily interoperate with each other
without operator assistance and expensive additional equipment that
translates from one vendor to another. It would be like having to make
sure all parties are on the same equipment (and network) when making a
telephone call.  A major factor in the inability of Telepresence systems
to work with each other is that there is no standard description of the
multiple streams that comprise the media flows.=20

=20

For example, in a multiple screen conference, the video and audio
streams sent from remote participants must be understood by receivers so
that they can be presented in a coherent and life-like manner. This
includes the ability to present the remote participants at their true
size for their apparent distance, while maintaining correct eye contact,
gesticular cues, and simultaneously providing a spatial audio sound
stage consistent with the video presentation.  The receiving device that
decides how to display the incoming information needs to understand a
number of variables such as the spatial position of the speaker, the
field of view of the cameras, the camera zoom, which media stream is
related to each of the displays, etc.=20

=20

Charter

The Telepresence Multi-Streams work item in DISPATCH is chartered to
define and specify for SIP-based systems the content of media
multi-stream messages and the way these will be transported.=20

=20

This work will provide a standard for the exchange of media semantic
information that will foster interoperable end stations and conference
bridges. It will specify  variables that describe the semantics of the
media streams and the recommended behavior to achieve interoperability.


=20

This requires considering current widely deployed use cases, such as
single and multiple screens, multi-point, Scalable Video Coding (SVC),
as well as cases that are expected to be implemented using the protocol
framework produced by this work item.  The methodology for describing
the variables must allow extensibility of the variables, since
telepresence is still a young technology and may have use cases that are
not currently considered.

=20

The work item will identify use cases, define requirements, and define a
method for describing and transporting information about multiple media
streams, including a specification of variables required to support the
use cases. This work item will consider the reuse of existing IETF
protocols and produce an architecture/protocol framework document
describing the protocols required to be implemented to support this
functionality.  The document will identify any enhancements required to
existing protocols as well as describing new protocol(s) for
interoperable multi-streams negotiation that may be required.

=20

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and
FECframe.

=20

Scope

The scope includes both systems that provide a fully immersive
experience, and systems that interwork with them and therefore need to
understand the same multiple stream semantics.

=20

=20

The focus of this work is on audio and video multiple streams.  Other
media types may be considered, however development of methodologies for
them is not within the scope of this work.

=20

Interoperation with standards compliant systems is required, such as
SIP-based video conferencing systems.  However, backwards compatibility
with existing non-standards compliant telepresence systems is not
required.

=20

=20

=20

The group will produce

- Requirements and use cases

=20

- Architectural Framework describing the protocols required to be
implemented to support this functionality and identifying existing
protocol enhancements and new protocol functionality required

=20

- Specification of a new protocol to support telepresence multi-streams
[if required]

=20

Goals and Milestones

Nov 2010=20

=20

Use Cases and Requirements to IESG as Informational RFC=20

Nov 2010

March 2011=20

Problem Statement to IESG as Informational RFC

Architecture to IESG as Informational RFC=20

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011=20

Submit protocol draft to IESG as Proposed Standard RFC=20

=20

=20

=20


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

=20


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Peter,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It&#8217;s certainly possible imo to explicitly state =
RTP. We
could put it on the list for changes, if there is no =
objection.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Allyn<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Peter =
Musgrave
[mailto:peter.musgrave@magorcorp.com] <br>
<b>Sent:</b> Wednesday, June 02, 2010 9:40 AM<br>
<b>To:</b> Allyn Romanow (allyn)<br>
<b>Cc:</b> dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen =
Jennings
(fluffy)<br>
<b>Subject:</b> Re: [dispatch] Updated charter/work description for =
telepresence
multi-streams<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hi,&nbsp;<o:p></o:p></p>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>I read this through. I think it describes the work =
well
enough at this point.&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Is it worth narrowing the scope of how media will =
be
transmitted? I assume it will be RTP but this not explicitly =
stated.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>Regards,<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>Peter =
Musgrave<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Wed, Jun 2, 2010 at 12:11 PM, Allyn Romanow =
(allyn) &lt;<a
href=3D"mailto:allyn@cisco.com">allyn@cisco.com</a>&gt; =
wrote:<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Folks,<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Here
is a version of the work description that addresses the comments made on =
the
first draft. <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The
changes include: <o:p></o:p></p>

<p>1.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Charter
section, qualification that we are working on SIP-based =
systems<o:p></o:p></p>

<p>2.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Charter
section, minor word smithing<o:p></o:p></p>

<p>3.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Charter
section, reference to RAI WG with relevant work<o:p></o:p></p>

<p>4.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Scope
section, clarification of interoperability requirements<o:p></o:p></p>

<p>5.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Scope
section, replacement of second paragraph regarding treatment of =
non-video
non-audio media types<o:p></o:p></p>

<p>6.<span =
style=3D'font-size:7.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Goals
and Milestones &#8211; addition of a Problem Statement =
draft<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Comments
welcome<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Allyn<o:p></=
o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;text-align:center'><b>Multi-streams for Telepresence Description of =
Work</b><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;text-align:center'><b>&nbsp;</b><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;text-align:center'><b>&nbsp;</b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>&nbsp;</b=
><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>Backgroun=
d</b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>One
branch of video conferencing has evolved that is focused on immersive
&#8220;being there&#8221; experience.&nbsp; Referred to in various ways =
such as
virtual conferencing, telepresence or media spaces, early systems were =
mainly
research projects or business systems with limited deployments.&nbsp; In =
recent
years telepresence systems have seen considerable market success.&nbsp;
Following the model of early systems, the first wave of commercial =
systems have
been typically located in specially designed single-purpose rooms with =
multiple
relatively large displays permitting life size image reproduction, =
multiple
cameras, encoders, decoders and microphones.&nbsp; These systems have =
several
important characteristics that are different from more traditional video
conferencing systems.&nbsp; <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The
first difference concerns controlling the visual viewpoint in order to =
improve
participant nonverbal communication. These systems preserve essential =
group
meeting characteristics such as eye contact, group gestures, seating =
order and
spatial audio by carefully orchestrating the miking and camera angles at =
each
of the sites . This is distinct from the more traditional approach where =
the
geometric relationship between media streams is not used to preserve
inter-stream communication aspects such as eye contact and group
dynamics.&nbsp; <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>A
second difference is manipulation of the environment to improve =
immersion.&nbsp;
With telepresence systems, cinematographic aspects of the local =
environment
reproduction are carefully planned including color, table shape, seating =
and
lighting so that when combined with large high quality displays, a =
strong sense
of a &quot;trompe l'oeil&quot; or &quot;being there&quot; immersive =
experience
is created.&nbsp; Typical video conference systems do not include these
considerations.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>As
telepresence video systems have become successful in the market, =
manufacturers
have started exploring delivery of the nonverbal communication and =
immersive
values of telepresence via smaller, less expensive and more flexible =
video
conferencing systems for a variety of venues, such as individual =
offices, homes
and kiosks. These are also&nbsp; telepresence systems, since the audio =
and
video quality is high enough to allow clear image reproduction for =
nonverbal
communication, they are able to send and receive multiple media streams, =
and
large coordinated multi image displays are available for immersive
installations.&nbsp;&nbsp; As the industry develops, the line between
telepresence and video conferencing may become blurred as nonverbal
communication and immersive installations become broadly =
available.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>Problem</=
b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Although
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323
suite, they cannot easily interoperate with each other without operator
assistance and expensive additional equipment that translates from one =
vendor
to another. It would be like having to make sure all parties are on the =
same
equipment (and network) when making a telephone call. &nbsp;A major =
factor in
the inability of Telepresence systems to work with each other is that =
there is
no standard description of the multiple streams that comprise the media =
flows. <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>For
example, in a multiple screen conference, the video and audio streams =
sent from
remote participants must be understood by receivers so that they can be
presented in a coherent and life-like manner. This includes the ability =
to
present the remote participants at their true size for their apparent =
distance,
while maintaining correct eye contact, gesticular cues, and =
simultaneously
providing a spatial audio sound stage consistent with the video
presentation.&nbsp; The receiving device that decides how to display the
incoming information needs to understand a number of variables such as =
the
spatial position of the speaker, the field of view of the cameras, the =
camera
zoom, which media stream is related to each of the displays, etc. =
<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>Charter</=
b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The
Telepresence Multi-Streams work item in DISPATCH is chartered to define =
and
specify for SIP-based systems the content of media multi-stream messages =
and
the way these will be transported. <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>This work will provide a standard for the exchange =
of
media semantic information that will foster interoperable end stations =
and
conference bridges. It will specify&nbsp; variables that describe the =
semantics
of the media streams and the recommended behavior to achieve
interoperability.&nbsp; <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>This requires considering current widely deployed =
use
cases, such as single and multiple screens, multi-point, Scalable Video =
Coding
(SVC), as well as cases that are expected to be implemented using the =
protocol
framework produced by this work item.&nbsp; The methodology for =
describing the
variables must allow extensibility of the variables, since telepresence =
is
still a young technology and may have use cases that are not currently
considered.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>The work item will identify use cases, define
requirements, and define a method for describing and transporting =
information
about multiple media streams, including a specification of variables =
required
to support the use cases. This work item will consider the reuse of =
existing
IETF protocols and produce an architecture/protocol framework document =
describing
the protocols required to be implemented to support this =
functionality.&nbsp;
The document will identify any enhancements required to existing =
protocols as
well as describing new protocol(s) for interoperable multi-streams =
negotiation
that may be required.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>Relevant work to be drawn upon has been done in =
XCON,
MMUSIC, AVT, and FECframe.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'><b>Scope</b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The
scope includes both systems that provide a fully immersive experience, =
and
systems that interwork with them and therefore need to understand the =
same
multiple stream semantics.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p>The focus of this work is on audio and video multiple streams.&nbsp; =
Other
media types may be considered, however development of methodologies for =
them is
not within the scope of this work.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>Interoperation with standards compliant systems is
required, such as SIP-based video conferencing systems. &nbsp;However,
backwards compatibility with existing non-standards compliant =
telepresence
systems is not required.<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'><b>&nbsp;</b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'color:#0070C0'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>The
group will produce</b><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>- Requirements and use cases<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>- Architectural Framework describing the protocols
required to be implemented to support this functionality and identifying
existing protocol enhancements and new protocol functionality =
required<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-autospace:none'>- Specification of a new protocol to support =
telepresence
multi-streams [if required]<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b>Goals
and Milestones</b><o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Nov
  2010 <o:p></o:p></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Use
  Cases and Requirements to IESG as Informational RFC <o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Nov
  2010<o:p></o:p></p>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>March
  2011 <o:p></o:p></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Problem
  Statement to IESG as Informational RFC<o:p></o:p></p>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Architecture=

  to IESG as Informational RFC <o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>March
  2011<o:p></o:p></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Revise
  Charter [IF new protocol is not required]<o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Nov
  2011 <o:p></o:p></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Submit
  protocol draft to IESG as Proposed Standard RFC <o:p></o:p></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><o:p>=
</o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</body>

</html>

------_=_NextPart_001_01CB0272.ED12F31F--

From fluffy@cisco.com  Wed Jun  2 11:30:24 2010
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To: Muthu ArulMozhi Perumal (mperumal) <mperumal@cisco.com>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTNReachability (VIPR)
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Good catch - we certainly want it to just be one of many ways. 

On Jun 1, 2010, at 12:12 PM, Muthu ArulMozhi Perumal (mperumal) wrote:

> |The goal of this working group is to allows people to use SIP to
> |federate over the the internet while still using phone numbers to 
> |identify the person they wish to communicate with.
> 
> Do we want to use phone numbers as the only way to identify persons or
> allow people to use phone numbers as one of the ways to identify
> persons?
> 
> If it is the later, should we reword "still using phone numbers to
> identify" as "still allowing phone numbers to identify"?
> 
> thanks,
> Muthu           
> 
> |-----Original Message-----
> |From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> |Of Cullen Jennings (fluffy)
> |Sent: Tuesday, June 01, 2010 10:25 PM
> |To: DISPATCH list
> |Subject: [dispatch] Charter Proposal: Verification Involving
> |PSTNReachability (VIPR)
> |
> |
> |I've been talking to a lot of people about the VIPR drafts  - here is a
> |first cut of a proposal for a WG that could do this. I'm sure the
> charter
> |proposal needs a bunch of work but I wanted to get the discussion
> rolling on
> |the list.
> |
> |Thanks, Cullen
> |
> |(PS - this is sent in my individual contributor role. Take all my posts
> |about VIPR to be in my individual role not my co-chair role)
> |
> |-------------------------------------------------------
> |
> |ViPR Charter Proposal (Version 0)
> |
> |WG Name:  Verification Involving PSTN Reachability (VIPR)
> |
> |There are two globally deployed address spaces for communications that
> more
> |than a billion people use on a daily basis. They are phone numbers and
> DNS
> |rooted address such as web servers and email addresses. The federation
> |design of SIP is primarily designed for email style addresses yet a
> large
> |percentage of SIP deployments primarily use phone numbers for
> identifying
> |users. The goal of this working group is to allows people to use SIP to
> |federate over the the internet while still using phone numbers to
> identify
> |the person they wish to communicate with.
> |
> |The VIPR WG will address this problem by developing a peer to peer
> based
> |approach to finding SIP domains that claim to be responsible for a
> given
> |phone number and the WG will design validation protocols to ensure a
> |reasonable likelihood that a given domain actually is responsible for
> the
> |phone number. One initial validation protocol will be based on a domain
> |being able to prove it received a particular phone call over the PSTN
> based
> |on both sides knowing the start and stop times of that call. Other
> |validation schemes, such as examining fingerprints or watermarking of
> PSTN
> |media, to show that a domain received a particular PSTN phone call may
> be
> |considered by the working group. To help mitigate SPAM over SIP issues,
> the
> |WG will define an token based authorization scheme so that domain using
> SIP
> |to federate can choose to check that incoming SIP calls are from a
> domain
> |that successfully validated a phone number.
> |
> |The problem statement and some possible starting points for solutions
> are
> |further desired in the following internet drafts which shall form the
> bases
> |of the WG documents:
> |draft-rosenberg-dispatch-vipr-overview
> |draft-rosenberg-dispatch-vipr-reload-usage
> |draft-rosenberg-dispatch-vipr-pvp
> |draft-rosenberg-dispatch-vipr-sip-antispam
> |
> |The working group will carefully coordinate with the security area, O&M
> |area, as well as the appropriate RAI WG including sipcore and p2psip.
> |
> |
> |_______________________________________________
> |dispatch mailing list
> |dispatch@ietf.org
> |https://www.ietf.org/mailman/listinfo/dispatch


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From petithug@acm.org  Wed Jun  2 11:31:38 2010
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 06/01/2010 09:55 AM, Cullen Jennings wrote:
> 
> I've been talking to a lot of people about the VIPR drafts  - here is a first cut of a proposal for a WG that could do this. I'm sure the charter proposal needs a bunch of work but I wanted to get the discussion rolling on the list. 
> 
> Thanks, Cullen 
> 
> (PS - this is sent in my individual contributor role. Take all my posts about VIPR to be in my individual role not my co-chair role)
> 
> -------------------------------------------------------
> 
> ViPR Charter Proposal (Version 0)
> 
> WG Name:  Verification Involving PSTN Reachability (VIPR)
> 
> There are two globally deployed address spaces for communications that more than a billion people use on a daily basis. They are phone numbers and DNS rooted address such as web servers and email addresses. The federation design of SIP is primarily designed for email style addresses yet a large percentage of SIP deployments primarily use phone numbers for identifying users. The goal of this working group is to allows people to use SIP to federate over the the internet while still using phone numbers to identify the person they wish to communicate with. 
> 
> The VIPR WG will address this problem by developing a peer to peer based approach to finding SIP domains that claim to be responsible for a given phone number and the WG will design validation protocols to ensure a reasonable likelihood that a given domain actually is responsible for the phone number. One initial validation protocol will be based on a domain being able to prove it received a particular phone call over the PSTN based on both sides knowing the start and stop times of that call. Other validation schemes, such as examining fingerprints or watermarking of PSTN media, to show that a domain received a particular PSTN phone call may be considered by the working group. To help mitigate SPAM over SIP issues, the WG will define an token based authorization scheme so that domain using SIP to federate can choose to check that incoming SIP calls are from a domain that successfully validated a phone number. 
> 
> The problem statement and some possible starting points for solutions are further desired in the following internet drafts which shall form the bases of the WG documents:
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam

VAP is not listed here.  Is it an oversight or is it because the WG will work
only on interoperability between ViPR servers?

> 
> The working group will carefully coordinate with the security area, O&M area, as well as the appropriate RAI WG including sipcore and p2psip. 

I think something should said about the fact that ViPR requires SIP over (D)TLS
and SRTP, as it is, IMO, an important characteristic of ViPR.

Also there is perhaps a need to define a minimal subset of SIP extensions and
media capabilities for ViPR.

Thanks.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From fluffy@cisco.com  Wed Jun  2 11:36:03 2010
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To: Peter Musgrave <peter.musgrave@magorcorp.com>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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I kept trying to write something about the relation of ViPR to ENUM but =
it kept coming out sounding like public ENUM had failed - which was not =
really what I wanted to say so I eventually deleted it. If someone had =
some good text about ENUM, would be nice to add it.=20

I view infrastructure and carrier ENUM as somewhat unrelated to this but =
public ENUM is related. However, ENUM talks about how to query the =
database, but it is somewhat silent on how data gets in the database. =
How does the database authorize that someone is authoritative for a =
number and can put write a record into the ENUM database. VIPR is more =
about that side of the problem.=20

And +1 to Paul's answers of the other questions ....

On Jun 1, 2010, at 12:28 PM, Peter Musgrave wrote:

> Hi Cullen,
>=20
> Does the charter need to say anything explicitly about not relying on
> any of the work which uses ENUMs in DNS as a way of associating SIP
> endpoints and IP addresses?
>=20
> AFAIK ViPR is about *using* a SIP call to determine a path to the
> endpoint (however that happens to work).
>=20
> Does the charter need to stipulate that the mapping from a PSTN number
> to a SIP device is a one-to-one mapping? Does there need to be text
> about how multiple endpoints which are behind a border device (and all
> appear as the same PSTN number) are to be handled?.
>=20
> ViPR so far describes an "edge-device" approach. Is the charter
> restricted to this approach?
>=20
> Thanks,
>=20
> Peter Musgrave
>=20
>=20
>=20
> On Tue, Jun 1, 2010 at 12:55 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
> >
> > I've been talking to a lot of people about the VIPR drafts  - here =
is a first cut of a proposal for a WG that could do this. I'm sure the =
charter proposal needs a bunch of work but I wanted to get the =
discussion rolling on the list.
> >
> > Thanks, Cullen
> >
> > (PS - this is sent in my individual contributor role. Take all my =
posts about VIPR to be in my individual role not my co-chair role)
> >
> > -------------------------------------------------------
> >
> > ViPR Charter Proposal (Version 0)
> >
> > WG Name:  Verification Involving PSTN Reachability (VIPR)
> >
> > There are two globally deployed address spaces for communications =
that more than a billion people use on a daily basis. They are phone =
numbers and DNS rooted address such as web servers and email addresses. =
The federation design of SIP is primarily designed for email style =
addresses yet a large percentage of SIP deployments primarily use phone =
numbers for identifying users. The goal of this working group is to =
allows people to use SIP to federate over the the internet while still =
using phone numbers to identify the person they wish to communicate =
with.
> >
> > The VIPR WG will address this problem by developing a peer to peer =
based approach to finding SIP domains that claim to be responsible for a =
given phone number and the WG will design validation protocols to ensure =
a reasonable likelihood that a given domain actually is responsible for =
the phone number. One initial validation protocol will be based on a =
domain being able to prove it received a particular phone call over the =
PSTN based on both sides knowing the start and stop times of that call. =
Other validation schemes, such as examining fingerprints or watermarking =
of PSTN media, to show that a domain received a particular PSTN phone =
call may be considered by the working group. To help mitigate SPAM over =
SIP issues, the WG will define an token based authorization scheme so =
that domain using SIP to federate can choose to check that incoming SIP =
calls are from a domain that successfully validated a phone number.
> >
> > The problem statement and some possible starting points for =
solutions are further desired in the following internet drafts which =
shall form the bases of the WG documents:
> > draft-rosenberg-dispatch-vipr-overview
> > draft-rosenberg-dispatch-vipr-reload-usage
> > draft-rosenberg-dispatch-vipr-pvp
> > draft-rosenberg-dispatch-vipr-sip-antispam
> >
> > The working group will carefully coordinate with the security area, =
O&M area, as well as the appropriate RAI WG including sipcore and =
p2psip.
> >
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
>=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From fluffy@cisco.com  Wed Jun  2 11:39:26 2010
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+1 to ALES or SALUD


From peter.musgrave@magorcorp.com  Wed Jun  2 11:50:35 2010
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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How about:

"The essential characteristic of ViPR is establishing authentication
by PSTN reachability and not by e.g. direct reference to ENUM
databases or other assertions of PSTN number ownership. Elements such
as public ENUM may be employed indirectly as part of assessing the
PSTN reachability but no direct interaction with ENUM will be
required. "

Peter

On Wed, Jun 2, 2010 at 2:35 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
> I kept trying to write something about the relation of ViPR to ENUM but i=
t kept coming out sounding like public ENUM had failed - which was not real=
ly what I wanted to say so I eventually deleted it. If someone had some goo=
d text about ENUM, would be nice to add it.
>
> I view infrastructure and carrier ENUM as somewhat unrelated to this but =
public ENUM is related. However, ENUM talks about how to query the database=
, but it is somewhat silent on how data gets in the database. How does the =
database authorize that someone is authoritative for a number and can put w=
rite a record into the ENUM database. VIPR is more about that side of the p=
roblem.
>
> And +1 to Paul's answers of the other questions ....
>
> On Jun 1, 2010, at 12:28 PM, Peter Musgrave wrote:
>
>> Hi Cullen,
>>
>> Does the charter need to say anything explicitly about not relying on
>> any of the work which uses ENUMs in DNS as a way of associating SIP
>> endpoints and IP addresses?
>>
>> AFAIK ViPR is about *using* a SIP call to determine a path to the
>> endpoint (however that happens to work).
>>
>> Does the charter need to stipulate that the mapping from a PSTN number
>> to a SIP device is a one-to-one mapping? Does there need to be text
>> about how multiple endpoints which are behind a border device (and all
>> appear as the same PSTN number) are to be handled?.
>>
>> ViPR so far describes an "edge-device" approach. Is the charter
>> restricted to this approach?
>>
>> Thanks,
>>
>> Peter Musgrave
>>
>>
>>
>> On Tue, Jun 1, 2010 at 12:55 PM, Cullen Jennings <fluffy@cisco.com> wrot=
e:
>> >
>> > I've been talking to a lot of people about the VIPR drafts =A0- here i=
s a first cut of a proposal for a WG that could do this. I'm sure the chart=
er proposal needs a bunch of work but I wanted to get the discussion rollin=
g on the list.
>> >
>> > Thanks, Cullen
>> >
>> > (PS - this is sent in my individual contributor role. Take all my post=
s about VIPR to be in my individual role not my co-chair role)
>> >
>> > -------------------------------------------------------
>> >
>> > ViPR Charter Proposal (Version 0)
>> >
>> > WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>> >
>> > There are two globally deployed address spaces for communications that=
 more than a billion people use on a daily basis. They are phone numbers an=
d DNS rooted address such as web servers and email addresses. The federatio=
n design of SIP is primarily designed for email style addresses yet a large=
 percentage of SIP deployments primarily use phone numbers for identifying =
users. The goal of this working group is to allows people to use SIP to fed=
erate over the the internet while still using phone numbers to identify the=
 person they wish to communicate with.
>> >
>> > The VIPR WG will address this problem by developing a peer to peer bas=
ed approach to finding SIP domains that claim to be responsible for a given=
 phone number and the WG will design validation protocols to ensure a reaso=
nable likelihood that a given domain actually is responsible for the phone =
number. One initial validation protocol will be based on a domain being abl=
e to prove it received a particular phone call over the PSTN based on both =
sides knowing the start and stop times of that call. Other validation schem=
es, such as examining fingerprints or watermarking of PSTN media, to show t=
hat a domain received a particular PSTN phone call may be considered by the=
 working group. To help mitigate SPAM over SIP issues, the WG will define a=
n token based authorization scheme so that domain using SIP to federate can=
 choose to check that incoming SIP calls are from a domain that successfull=
y validated a phone number.
>> >
>> > The problem statement and some possible starting points for solutions =
are further desired in the following internet drafts which shall form the b=
ases of the WG documents:
>> > draft-rosenberg-dispatch-vipr-overview
>> > draft-rosenberg-dispatch-vipr-reload-usage
>> > draft-rosenberg-dispatch-vipr-pvp
>> > draft-rosenberg-dispatch-vipr-sip-antispam
>> >
>> > The working group will carefully coordinate with the security area, O&=
M area, as well as the appropriate RAI WG including sipcore and p2psip.
>> >
>> >
>> > _______________________________________________
>> > dispatch mailing list
>> > dispatch@ietf.org
>> > https://www.ietf.org/mailman/listinfo/dispatch
>> >
>>
>
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
>
>

From fluffy@cisco.com  Wed Jun  2 11:56:42 2010
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Subject: Re: [dispatch] Charter proposal for drafts related to the tel URI
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Can we identify the complete list of  semantic information this WG would =
cover? Some working groups have tried to charter with just "some =
information" and it is so vague that it really slows down the WG.=20

Then can we look at at least one real world use case for each type of =
information - Take the pay phone case for example, what is the use case =
where someone needs to know the other end is a pay phone and what do =
they do differently.


On May 28, 2010, at 7:13 AM, Patel, Milan wrote:

> Hello,
> =20
> Attached is a charter proposal for a WG to be formed for work =
pertaining to extensions to the tel URI =96 specifically, parameters for =
carrying the DAI, CPC, OLI and RN required for primarily interworking =
scenarios to ISUP/PSTN.
> Feedback is appreciated.
> Regards,
> Milan
> =20
> =20
> From: Patel, Milan=20
> Sent: Tuesday, May 25, 2010 11:16 AM
> To: Cullen Jennings; 'mary.ietf.barnes@gmail.com'; Robert Sparks; =
'Gonzalo Camarillo'
> Cc: R.Jesske@telekom.de; 'Sumanth Channabasappa'
> Subject: Charter proposal
> =20
> Hi all,
> =20
> Attached is a charter proposal for a WG to be formed to handle =
Internet Drafts pertaining to the Tel URI. Specifically these are the =
DAI draft and the CPC/OLI draft and some further work on the RN =
parameter.
> Feedback is welcome.
> =20
> Kind regards,
> Milan
> <XXXX WG charter proposal.txt>


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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Subject: Re: [dispatch] Multi-streams for telepresence description of work
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On May 19, 2010, at 5:19 AM, stephen botzko wrote:

> Allyn and I co-chair the IMTC multi-streaming group.  The IMTC would =
be delighted if the IETF takes on this work item.

And at some time in the past, the RAI ADs told IMTC that they would be =
delighted to see the IMTC bring this type of work to the IETF.


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A comment on the following sentence in the updated charter:

=20

"This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  "

=20

Addressing use cases such as single and multiple screen systems, as well =
as both point-to-point and multi-point calling is great.  However I do =
not understand why a specific video codec or coding technique has been =
added to the same context.  The charter shouldn't recommend or limit the =
work for any specific codec (it should be codec agnostic),  and I =
suggest the charter should be changed to comply with this. =20

=20

-h

=20

________________________________

Fra: dispatch-bounces@ietf.org p=E5 vegne av Allyn Romanow (allyn)
Sendt: on 02.06.2010 18:11
Til: dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings =
(fluffy)
Emne: [dispatch] Updated charter/work description for =
telepresencemulti-streams

Folks,

Here is a version of the work description that addresses the comments =
made on the first draft.=20

The changes include:=20

1.       Charter section, qualification that we are working on SIP-based =
systems

2.       Charter section, minor word smithing

3.       Charter section, reference to RAI WG with relevant work

4.       Scope section, clarification of interoperability requirements

5.       Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types

6.       Goals and Milestones - addition of a Problem Statement draft

=20

=20

Comments welcome

=20

Allyn

=20

Multi-streams for Telepresence Description of Work

=20

=20

=20

Background

One branch of video conferencing has evolved that is focused on =
immersive "being there" experience.  Referred to in various ways such as =
virtual conferencing, telepresence or media spaces, early systems were =
mainly research projects or business systems with limited deployments.  =
In recent years telepresence systems have seen considerable market =
success.  Following the model of early systems, the first wave of =
commercial systems have been typically located in specially designed =
single-purpose rooms with multiple relatively large displays permitting =
life size image reproduction, multiple cameras, encoders, decoders and =
microphones.  These systems have several important characteristics that =
are different from more traditional video conferencing systems. =20

=20

The first difference concerns controlling the visual viewpoint in order =
to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group =
gestures, seating order and spatial audio by carefully orchestrating the =
miking and camera angles at each of the sites . This is distinct from =
the more traditional approach where the geometric relationship between =
media streams is not used to preserve inter-stream communication aspects =
such as eye contact and group dynamics. =20

=20

A second difference is manipulation of the environment to improve =
immersion.  With telepresence systems, cinematographic aspects of the =
local environment reproduction are carefully planned including color, =
table shape, seating and lighting so that when combined with large high =
quality displays, a strong sense of a "trompe l'oeil" or "being there" =
immersive experience is created.  Typical video conference systems do =
not include these considerations.

=20

As telepresence video systems have become successful in the market, =
manufacturers have started exploring delivery of the nonverbal =
communication and immersive values of telepresence via smaller, less =
expensive and more flexible video conferencing systems for a variety of =
venues, such as individual offices, homes and kiosks. These are also  =
telepresence systems, since the audio and video quality is high enough =
to allow clear image reproduction for nonverbal communication, they are =
able to send and receive multiple media streams, and large coordinated =
multi image displays are available for immersive installations.   As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.

=20

Problem

Although telepresence systems are based on open standards such as RTP, =
SIP, H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call.  A major factor in the inability of Telepresence systems =
to work with each other is that there is no standard description of the =
multiple streams that comprise the media flows.=20

=20

For example, in a multiple screen conference, the video and audio =
streams sent from remote participants must be understood by receivers so =
that they can be presented in a coherent and life-like manner. This =
includes the ability to present the remote participants at their true =
size for their apparent distance, while maintaining correct eye contact, =
gesticular cues, and simultaneously providing a spatial audio sound =
stage consistent with the video presentation.  The receiving device that =
decides how to display the incoming information needs to understand a =
number of variables such as the spatial position of the speaker, the =
field of view of the cameras, the camera zoom, which media stream is =
related to each of the displays, etc.=20

=20

Charter

The Telepresence Multi-Streams work item in DISPATCH is chartered to =
define and specify for SIP-based systems the content of media =
multi-stream messages and the way these will be transported.=20

=20

This work will provide a standard for the exchange of media semantic =
information that will foster interoperable end stations and conference =
bridges. It will specify  variables that describe the semantics of the =
media streams and the recommended behavior to achieve interoperability.  =


=20

This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  The methodology for describing =
the variables must allow extensibility of the variables, since =
telepresence is still a young technology and may have use cases that are =
not currently considered.

=20

The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media =
streams, including a specification of variables required to support the =
use cases. This work item will consider the reuse of existing IETF =
protocols and produce an architecture/protocol framework document =
describing the protocols required to be implemented to support this =
functionality.  The document will identify any enhancements required to =
existing protocols as well as describing new protocol(s) for =
interoperable multi-streams negotiation that may be required.

=20

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.

=20

Scope

The scope includes both systems that provide a fully immersive =
experience, and systems that interwork with them and therefore need to =
understand the same multiple stream semantics.

=20

=20

The focus of this work is on audio and video multiple streams.  Other =
media types may be considered, however development of methodologies for =
them is not within the scope of this work.

=20

Interoperation with standards compliant systems is required, such as =
SIP-based video conferencing systems.  However, backwards compatibility =
with existing non-standards compliant telepresence systems is not =
required.

=20

=20

=20

The group will produce

- Requirements and use cases

=20

- Architectural Framework describing the protocols required to be =
implemented to support this functionality and identifying existing =
protocol enhancements and new protocol functionality required

=20

- Specification of a new protocol to support telepresence multi-streams =
[if required]

=20

Goals and Milestones

Nov 2010=20

=20

Use Cases and Requirements to IESG as Informational RFC=20

Nov 2010

March 2011=20

Problem Statement to IESG as Informational RFC

Architecture to IESG as Informational RFC=20

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011=20

Submit protocol draft to IESG as Proposed Standard RFC=20

=20

=20

=20


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</o:shapelayout></xml><![endif]--></head><body lang=3DNO-BOK link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:12.0pt'>A comment on the following =
sentence in the updated charter:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>&#8220;This requires considering current =
widely deployed use cases, such as single and multiple screens, =
multi-point, Scalable Video Coding (SVC), as well as cases that are =
expected to be implemented using the protocol framework produced by this =
work item.&nbsp; &#8221;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'>Addressing =
use cases such as single and multiple screen systems, as well as both =
point-to-point and multi-point calling is great. &nbsp;However I do not =
understand why a specific video codec or coding technique has been added =
to the same context.&nbsp; The charter shouldn&#8217;t recommend or =
limit the work for any specific codec (it should be codec =
agnostic),&nbsp; and I suggest the charter should be changed to comply =
with this.&nbsp; <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:12.0pt;color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'>-h<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div =
class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><span =
lang=3DEN-US style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><hr size=3D2 width=3D"100%" =
align=3Dcenter></span></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fra:</span><=
/b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
dispatch-bounces@ietf.org p=E5 vegne av Allyn Romanow =
(allyn)<br><b>Sendt:</b> on 02.06.2010 18:11<br><b>Til:</b> =
dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings =
(fluffy)<br><b>Emne:</b> [dispatch] Updated charter/work description for =
telepresencemulti-streams</span><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p><div><p class=3DMsoNormal><span =
lang=3DEN-US>Folks,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Here is a version of the work description that addresses =
the comments made on the first draft. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>The changes include: =
<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>1.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, qualification that we are working on =
SIP-based systems<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>2.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, minor word =
smithing<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>3.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, reference to RAI WG with relevant =
work<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>4.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Scope section, clarification of interoperability =
requirements<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>5.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types<o:p></o:p></span></p><p =
class=3DMsoListParagraph style=3D'text-indent:-18.0pt'><span =
lang=3DEN-US>6.</span><span lang=3DEN-US =
style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Goals and Milestones &#8211; addition of a Problem =
Statement draft<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Comments welcome<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Allyn<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Multi-streams =
for Telepresence Description of Work</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal =
align=3Dcenter style=3D'text-align:center'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Background</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>One branch of =
video conferencing has evolved that is focused on immersive &#8220;being =
there&#8221; experience.&nbsp; Referred to in various ways such as =
virtual conferencing, telepresence or media spaces, early systems were =
mainly research projects or business systems with limited =
deployments.&nbsp; In recent years telepresence systems have seen =
considerable market success.&nbsp; Following the model of early systems, =
the first wave of commercial systems have been typically located in =
specially designed single-purpose rooms with multiple relatively large =
displays permitting life size image reproduction, multiple cameras, =
encoders, decoders and microphones.&nbsp; These systems have several =
important characteristics that are different from more traditional video =
conferencing systems.&nbsp; </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The first =
difference concerns controlling the visual viewpoint in order to improve =
participant nonverbal communication. These systems preserve essential =
group meeting characteristics such as eye contact, group gestures, =
seating order and spatial audio by carefully orchestrating the miking =
and camera angles at each of the sites . This is distinct from the more =
traditional approach where the geometric relationship between media =
streams is not used to preserve inter-stream communication aspects such =
as eye contact and group dynamics.&nbsp; </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>A second =
difference is manipulation of the environment to improve =
immersion.&nbsp; With telepresence systems, cinematographic aspects of =
the local environment reproduction are carefully planned including =
color, table shape, seating and lighting so that when combined with =
large high quality displays, a strong sense of a &quot;trompe =
l'oeil&quot; or &quot;being there&quot; immersive experience is =
created.&nbsp; Typical video conference systems do not include these =
considerations.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>As telepresence video systems =
have become successful in the market, manufacturers have started =
exploring delivery of the nonverbal communication and immersive values =
of telepresence via smaller, less expensive and more flexible video =
conferencing systems for a variety of venues, such as individual =
offices, homes and kiosks. These are also&nbsp; telepresence systems, =
since the audio and video quality is high enough to allow clear image =
reproduction for nonverbal communication, they are able to send and =
receive multiple media streams, and large coordinated multi image =
displays are available for immersive installations.&nbsp;&nbsp; As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Problem</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Although =
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call. &nbsp;A major factor in the inability of Telepresence =
systems to work with each other is that there is no standard description =
of the multiple streams that comprise the media flows. </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>For example, in =
a multiple screen conference, the video and audio streams sent from =
remote participants must be understood by receivers so that they can be =
presented in a coherent and life-like manner. This includes the ability =
to present the remote participants at their true size for their apparent =
distance, while maintaining correct eye contact, gesticular cues, and =
simultaneously providing a spatial audio sound stage consistent with the =
video presentation.&nbsp; The receiving device that decides how to =
display the incoming information needs to understand a number of =
variables such as the spatial position of the speaker, the field of view =
of the cameras, the camera zoom, which media stream is related to each =
of the displays, etc. </span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Charter</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The Telepresence =
Multi-Streams work item in DISPATCH is chartered to define and specify =
for SIP-based systems the content of media multi-stream messages and the =
way these will be transported. </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>This work will =
provide a standard for the exchange of media semantic information that =
will foster interoperable end stations and conference bridges. It will =
specify&nbsp; variables that describe the semantics of the media streams =
and the recommended behavior to achieve interoperability.&nbsp; =
</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>This requires considering =
current widely deployed use cases, such as single and multiple screens, =
multi-point, Scalable Video Coding (SVC), as well as cases that are =
expected to be implemented using the protocol framework produced by this =
work item.&nbsp; The methodology for describing the variables must allow =
extensibility of the variables, since telepresence is still a young =
technology and may have use cases that are not currently =
considered.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>The work item will identify =
use cases, define requirements, and define a method for describing and =
transporting information about multiple media streams, including a =
specification of variables required to support the use cases. This work =
item will consider the reuse of existing IETF protocols and produce an =
architecture/protocol framework document describing the protocols =
required to be implemented to support this functionality.&nbsp; The =
document will identify any enhancements required to existing protocols =
as well as describing new protocol(s) for interoperable multi-streams =
negotiation that may be required.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Relevant work to =
be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Scope</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The scope =
includes both systems that provide a fully immersive experience, and =
systems that interwork with them and therefore need to understand the =
same multiple stream semantics.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'>The focus of =
this work is on audio and video multiple streams.&nbsp; Other media =
types may be considered, however development of methodologies for them =
is not within the scope of this work.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Interoperation =
with standards compliant systems is required, such as SIP-based video =
conferencing systems. &nbsp;However, backwards compatibility with =
existing non-standards compliant telepresence systems is not =
required.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>The group will =
produce</span></b><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>- Requirements and use =
cases</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>- Architectural Framework =
describing the protocols required to be implemented to support this =
functionality and identifying existing protocol enhancements and new =
protocol functionality required</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>- Specification =
of a new protocol to support telepresence multi-streams [if =
required]</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Goals and =
Milestones</span></b><span lang=3DEN-US><o:p></o:p></span></p><table =
class=3DMsoNormalTable border=3D0 cellpadding=3D0><tr><td width=3D96 =
style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2010 </span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Use Cases and Requirements to =
IESG as Informational RFC </span><o:p></o:p></p></td></tr><tr><td =
width=3D96 style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2010</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011 =
</span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Problem Statement to IESG as =
Informational RFC</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Architecture to IESG as =
Informational RFC </span><o:p></o:p></p></td></tr><tr><td width=3D96 =
style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>March =
2011</span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Revise Charter [IF new =
protocol is not required]</span><o:p></o:p></p></td></tr><tr><td =
width=3D96 style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2011 </span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Submit protocol draft to IESG =
as Proposed Standard RFC </span><o:p></o:p></p></td></tr></table><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div></body></html>
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A comment on the following sentence in the updated charter:

=20

"This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  "

=20

Addressing use cases such as single and multiple screen systems, as well =
as both point-to-point and multi-point calling is great.  However I do =
not understand why a specific video codec or coding technique has been =
added to the same context.  The charter shouldn't recommend or limit the =
work for any specific codec (it should be codec agnostic),  and I =
suggest the charter should be changed to comply with this. =20

=20

-h

=20

________________________________

Fra: dispatch-bounces@ietf.org p=E5 vegne av Allyn Romanow (allyn)
Sendt: on 02.06.2010 18:11
Til: dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings =
(fluffy)
Emne: [dispatch] Updated charter/work description for =
telepresencemulti-streams

Folks,

Here is a version of the work description that addresses the comments =
made on the first draft.=20

The changes include:=20

1.       Charter section, qualification that we are working on SIP-based =
systems

2.       Charter section, minor word smithing

3.       Charter section, reference to RAI WG with relevant work

4.       Scope section, clarification of interoperability requirements

5.       Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types

6.       Goals and Milestones - addition of a Problem Statement draft

=20

=20

Comments welcome

=20

Allyn

=20

Multi-streams for Telepresence Description of Work

=20

=20

=20

Background

One branch of video conferencing has evolved that is focused on =
immersive "being there" experience.  Referred to in various ways such as =
virtual conferencing, telepresence or media spaces, early systems were =
mainly research projects or business systems with limited deployments.  =
In recent years telepresence systems have seen considerable market =
success.  Following the model of early systems, the first wave of =
commercial systems have been typically located in specially designed =
single-purpose rooms with multiple relatively large displays permitting =
life size image reproduction, multiple cameras, encoders, decoders and =
microphones.  These systems have several important characteristics that =
are different from more traditional video conferencing systems. =20

=20

The first difference concerns controlling the visual viewpoint in order =
to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group =
gestures, seating order and spatial audio by carefully orchestrating the =
miking and camera angles at each of the sites . This is distinct from =
the more traditional approach where the geometric relationship between =
media streams is not used to preserve inter-stream communication aspects =
such as eye contact and group dynamics. =20

=20

A second difference is manipulation of the environment to improve =
immersion.  With telepresence systems, cinematographic aspects of the =
local environment reproduction are carefully planned including color, =
table shape, seating and lighting so that when combined with large high =
quality displays, a strong sense of a "trompe l'oeil" or "being there" =
immersive experience is created.  Typical video conference systems do =
not include these considerations.

=20

As telepresence video systems have become successful in the market, =
manufacturers have started exploring delivery of the nonverbal =
communication and immersive values of telepresence via smaller, less =
expensive and more flexible video conferencing systems for a variety of =
venues, such as individual offices, homes and kiosks. These are also  =
telepresence systems, since the audio and video quality is high enough =
to allow clear image reproduction for nonverbal communication, they are =
able to send and receive multiple media streams, and large coordinated =
multi image displays are available for immersive installations.   As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.

=20

Problem

Although telepresence systems are based on open standards such as RTP, =
SIP, H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call.  A major factor in the inability of Telepresence systems =
to work with each other is that there is no standard description of the =
multiple streams that comprise the media flows.=20

=20

For example, in a multiple screen conference, the video and audio =
streams sent from remote participants must be understood by receivers so =
that they can be presented in a coherent and life-like manner. This =
includes the ability to present the remote participants at their true =
size for their apparent distance, while maintaining correct eye contact, =
gesticular cues, and simultaneously providing a spatial audio sound =
stage consistent with the video presentation.  The receiving device that =
decides how to display the incoming information needs to understand a =
number of variables such as the spatial position of the speaker, the =
field of view of the cameras, the camera zoom, which media stream is =
related to each of the displays, etc.=20

=20

Charter

The Telepresence Multi-Streams work item in DISPATCH is chartered to =
define and specify for SIP-based systems the content of media =
multi-stream messages and the way these will be transported.=20

=20

This work will provide a standard for the exchange of media semantic =
information that will foster interoperable end stations and conference =
bridges. It will specify  variables that describe the semantics of the =
media streams and the recommended behavior to achieve interoperability.  =


=20

This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  The methodology for describing =
the variables must allow extensibility of the variables, since =
telepresence is still a young technology and may have use cases that are =
not currently considered.

=20

The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media =
streams, including a specification of variables required to support the =
use cases. This work item will consider the reuse of existing IETF =
protocols and produce an architecture/protocol framework document =
describing the protocols required to be implemented to support this =
functionality.  The document will identify any enhancements required to =
existing protocols as well as describing new protocol(s) for =
interoperable multi-streams negotiation that may be required.

=20

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.

=20

Scope

The scope includes both systems that provide a fully immersive =
experience, and systems that interwork with them and therefore need to =
understand the same multiple stream semantics.

=20

=20

The focus of this work is on audio and video multiple streams.  Other =
media types may be considered, however development of methodologies for =
them is not within the scope of this work.

=20

Interoperation with standards compliant systems is required, such as =
SIP-based video conferencing systems.  However, backwards compatibility =
with existing non-standards compliant telepresence systems is not =
required.

=20

=20

=20

The group will produce

- Requirements and use cases

=20

- Architectural Framework describing the protocols required to be =
implemented to support this functionality and identifying existing =
protocol enhancements and new protocol functionality required

=20

- Specification of a new protocol to support telepresence multi-streams =
[if required]

=20

Goals and Milestones

Nov 2010=20

=20

Use Cases and Requirements to IESG as Informational RFC=20

Nov 2010

March 2011=20

Problem Statement to IESG as Informational RFC

Architecture to IESG as Informational RFC=20

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011=20

Submit protocol draft to IESG as Proposed Standard RFC=20

=20

=20

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:12.0pt'>A comment on the following =
sentence in the updated charter:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>&#8220;This requires considering current =
widely deployed use cases, such as single and multiple screens, =
multi-point, Scalable Video Coding (SVC), as well as cases that are =
expected to be implemented using the protocol framework produced by this =
work item.=A0 &#8221;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'>Addressing =
use cases such as single and multiple screen systems, as well as both =
point-to-point and multi-point calling is great. =A0However I do not =
understand why a specific video codec or coding technique has been added =
to the same context.&nbsp; The charter shouldn&#8217;t recommend or =
limit the work for any specific codec (it should be codec =
agnostic),&nbsp; and I suggest the charter should be changed to comply =
with this.&nbsp; <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:12.0pt;color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'>-h<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div =
class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><span =
lang=3DEN-US style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><hr size=3D2 width=3D"100%" =
align=3Dcenter></span></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fra:</span><=
/b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
dispatch-bounces@ietf.org p=E5 vegne av Allyn Romanow =
(allyn)<br><b>Sendt:</b> on 02.06.2010 18:11<br><b>Til:</b> =
dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings =
(fluffy)<br><b>Emne:</b> [dispatch] Updated charter/work description for =
telepresencemulti-streams</span><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p><div><p class=3DMsoNormal><span =
lang=3DEN-US>Folks,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Here is a version of the work description that addresses =
the comments made on the first draft. <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>The changes include: =
<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>1.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, qualification that we are working on =
SIP-based systems<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>2.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, minor word =
smithing<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>3.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Charter section, reference to RAI WG with relevant =
work<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>4.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Scope section, clarification of interoperability =
requirements<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt'><span lang=3DEN-US>5.</span><span =
lang=3DEN-US style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types<o:p></o:p></span></p><p =
class=3DMsoListParagraph style=3D'text-indent:-18.0pt'><span =
lang=3DEN-US>6.</span><span lang=3DEN-US =
style=3D'font-size:7.0pt;font-family:"Times New =
Roman","serif"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><span =
lang=3DEN-US>Goals and Milestones &#8211; addition of a Problem =
Statement draft<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Comments welcome<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Allyn<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Multi-streams =
for Telepresence Description of Work</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal =
align=3Dcenter style=3D'text-align:center'><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Background</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>One branch of =
video conferencing has evolved that is focused on immersive &#8220;being =
there&#8221; experience.&nbsp; Referred to in various ways such as =
virtual conferencing, telepresence or media spaces, early systems were =
mainly research projects or business systems with limited =
deployments.&nbsp; In recent years telepresence systems have seen =
considerable market success.&nbsp; Following the model of early systems, =
the first wave of commercial systems have been typically located in =
specially designed single-purpose rooms with multiple relatively large =
displays permitting life size image reproduction, multiple cameras, =
encoders, decoders and microphones.&nbsp; These systems have several =
important characteristics that are different from more traditional video =
conferencing systems.&nbsp; </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The first =
difference concerns controlling the visual viewpoint in order to improve =
participant nonverbal communication. These systems preserve essential =
group meeting characteristics such as eye contact, group gestures, =
seating order and spatial audio by carefully orchestrating the miking =
and camera angles at each of the sites . This is distinct from the more =
traditional approach where the geometric relationship between media =
streams is not used to preserve inter-stream communication aspects such =
as eye contact and group dynamics.&nbsp; </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>A second =
difference is manipulation of the environment to improve =
immersion.&nbsp; With telepresence systems, cinematographic aspects of =
the local environment reproduction are carefully planned including =
color, table shape, seating and lighting so that when combined with =
large high quality displays, a strong sense of a &quot;trompe =
l'oeil&quot; or &quot;being there&quot; immersive experience is =
created.&nbsp; Typical video conference systems do not include these =
considerations.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>As telepresence video systems =
have become successful in the market, manufacturers have started =
exploring delivery of the nonverbal communication and immersive values =
of telepresence via smaller, less expensive and more flexible video =
conferencing systems for a variety of venues, such as individual =
offices, homes and kiosks. These are also&nbsp; telepresence systems, =
since the audio and video quality is high enough to allow clear image =
reproduction for nonverbal communication, they are able to send and =
receive multiple media streams, and large coordinated multi image =
displays are available for immersive installations.&nbsp;&nbsp; As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Problem</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Although =
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call. &nbsp;A major factor in the inability of Telepresence =
systems to work with each other is that there is no standard description =
of the multiple streams that comprise the media flows. </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>For example, in =
a multiple screen conference, the video and audio streams sent from =
remote participants must be understood by receivers so that they can be =
presented in a coherent and life-like manner. This includes the ability =
to present the remote participants at their true size for their apparent =
distance, while maintaining correct eye contact, gesticular cues, and =
simultaneously providing a spatial audio sound stage consistent with the =
video presentation.&nbsp; The receiving device that decides how to =
display the incoming information needs to understand a number of =
variables such as the spatial position of the speaker, the field of view =
of the cameras, the camera zoom, which media stream is related to each =
of the displays, etc. </span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Charter</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The Telepresence =
Multi-Streams work item in DISPATCH is chartered to define and specify =
for SIP-based systems the content of media multi-stream messages and the =
way these will be transported. </span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>This work will =
provide a standard for the exchange of media semantic information that =
will foster interoperable end stations and conference bridges. It will =
specify&nbsp; variables that describe the semantics of the media streams =
and the recommended behavior to achieve interoperability.&nbsp; =
</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>This requires considering =
current widely deployed use cases, such as single and multiple screens, =
multi-point, Scalable Video Coding (SVC), as well as cases that are =
expected to be implemented using the protocol framework produced by this =
work item.&nbsp; The methodology for describing the variables must allow =
extensibility of the variables, since telepresence is still a young =
technology and may have use cases that are not currently =
considered.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>The work item will identify =
use cases, define requirements, and define a method for describing and =
transporting information about multiple media streams, including a =
specification of variables required to support the use cases. This work =
item will consider the reuse of existing IETF protocols and produce an =
architecture/protocol framework document describing the protocols =
required to be implemented to support this functionality.&nbsp; The =
document will identify any enhancements required to existing protocols =
as well as describing new protocol(s) for interoperable multi-streams =
negotiation that may be required.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Relevant work to =
be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Scope</span></b><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>The scope =
includes both systems that provide a fully immersive experience, and =
systems that interwork with them and therefore need to understand the =
same multiple stream semantics.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'>The focus of =
this work is on audio and video multiple streams.&nbsp; Other media =
types may be considered, however development of methodologies for them =
is not within the scope of this work.</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>Interoperation =
with standards compliant systems is required, such as SIP-based video =
conferencing systems. &nbsp;However, backwards compatibility with =
existing non-standards compliant telepresence systems is not =
required.</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>The group will =
produce</span></b><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>- Requirements and use =
cases</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>- Architectural Framework =
describing the protocols required to be implemented to support this =
functionality and identifying existing protocol enhancements and new =
protocol functionality required</span><span =
lang=3DEN-US><o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-family:"Arial","sans-serif"'>- Specification =
of a new protocol to support telepresence multi-streams [if =
required]</span><span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif"'>Goals and =
Milestones</span></b><span lang=3DEN-US><o:p></o:p></span></p><table =
class=3DMsoNormalTable border=3D0 cellpadding=3D0><tr><td width=3D96 =
style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2010 </span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Use Cases and Requirements to =
IESG as Informational RFC </span><o:p></o:p></p></td></tr><tr><td =
width=3D96 style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2010</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011 =
</span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Problem Statement to IESG as =
Informational RFC</span><o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Architecture to IESG as =
Informational RFC </span><o:p></o:p></p></td></tr><tr><td width=3D96 =
style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>March =
2011</span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Revise Charter [IF new =
protocol is not required]</span><o:p></o:p></p></td></tr><tr><td =
width=3D96 style=3D'width:72.0pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Nov =
2011 </span><o:p></o:p></p></td><td width=3D381 =
style=3D'width:285.75pt;padding:.75pt .75pt .75pt .75pt'><p =
class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Submit protocol draft to IESG =
as Proposed Standard RFC </span><o:p></o:p></p></td></tr></table><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p></div></div></body></html>
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From: Jean-Francois Mule <jf.mule@cablelabs.com>
To: Cullen Jennings <fluffy@cisco.com>, DISPATCH list <dispatch@ietf.org>
Date: Wed, 2 Jun 2010 14:59:53 -0600
Thread-Topic: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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Hi,

   I support this proposed work to be taken by IETF under a new working gro=
up.  There are certainly concrete deliverables that would merit IETF consen=
sus to spur multi-vendor interoperability.

  A few comments below, mostly to help bash the proposed charter.

Jean-Francois.

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
> On Behalf Of Cullen Jennings
> Sent: Tuesday, June 01, 2010 10:55 AM
> To: DISPATCH list
> Subject: [dispatch] Charter Proposal: Verification Involving PSTN
> Reachability (VIPR)
>=20
>=20
> I've been talking to a lot of people about the VIPR drafts  - here
> is a first cut of a proposal for a WG that could do this. I'm sure
> the charter proposal needs a bunch of work but I wanted to get the
> discussion rolling on the list.
>=20
> Thanks, Cullen
>=20
> (PS - this is sent in my individual contributor role. Take all my
> posts about VIPR to be in my individual role not my co-chair role)
>=20
> -------------------------------------------------------
>=20
> ViPR Charter Proposal (Version 0)
>=20
> WG Name:  Verification Involving PSTN Reachability (VIPR)
>=20
> There are two globally deployed address spaces for communications
> that more than a billion people use on a daily basis. They are
> phone numbers and DNS rooted address such as web servers and email
> addresses. The federation design of SIP is primarily designed for
                 ^^^^^^^^^^^^^^^^^^^^^^^^
Given some folks talk about SIP federations in the context of speermint (ht=
tp://tools.ietf.org/html/rfc5486#section-5) and I don't think this is what =
you mean here, this choice of terms may not be the best.  As much as some f=
olks disagree on the worthiness of the output of speermint, at least SIP fe=
derations are described there.  Would recommend clarifying this somehow.
=20
> email style addresses yet a large percentage of SIP deployments
> primarily use phone numbers for identifying users. The goal of this
> working group is to allows people to use SIP to federate over the
                                       ^^^^^^^^^^^^^^^^^^^
my understanding is that those people would not only use SIP but some other=
 smarts to put some stuff in DHTs to enable PSTN reachability verification.=
 This implies that the solution solely relies on SIP.  May want to expand o=
r clarify, or else qualify the sentence further.

> the internet while still using phone numbers to identify the person
> they wish to communicate with.
>=20


> The VIPR WG will address this problem by developing a peer to peer
> based approach to finding SIP domains that claim to be responsible
                            ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
That may open a rathole: what does it mean to be responsible for a given ph=
one number (worst case, Rich Shockey will chime in and will muddy this more=
, or John Elwell will claim this should have been in speermint but it canno=
t be found there...).

Do you mean any SIP Service Provider (http://tools.ietf.org/html/rfc5486#se=
ction-3.9) on the signaling path of a SIP invite or dialog-initiating reque=
st for that telephone number?  Do you mean the SIP domain responsible for t=
he last hop before the SIP UA? Both?  Or is it the latter with the pre-requ=
isite that PSTN connectivity is required somehow?

> for a given phone number and the WG will design validation
> protocols to ensure a reasonable likelihood that a given domain
> actually is responsible for the phone number. One initial
> validation protocol will be based on a domain being able to prove
> it received a particular phone call over the PSTN based on both
> sides knowing the start and stop times of that call. Other
This last sentence is advocating one specific solution. =20
I think the VIPR drafts are powerful and this may be part of a working solu=
tion.  But at this stage in the WG charter definition, would it be preferab=
le to say something like:
	Some validation protocols may be based on additional knowledge gathered ar=
ound a SIP call, for example, the ability to prove a call was received over=
 the PSTN based on start and stop times.

It leave things open and welcomes other proposals.  It also does not preclu=
de the validation protocol you have documented.

> validation schemes, such as examining fingerprints or watermarking
> of PSTN media, to show that a domain received a particular PSTN
> phone call may be considered by the working group.=20
These are additional examples that must be part of the same bucket as the o=
ne before imo.  Currently it reads that PSTN+start+stop is the initial one,=
 others may be considered later. =20
I have no preference and like the elegance of the approach in some the VIPR=
 drafts.  But the charter should make clear that the WG will decide this.

> To help mitigate
> SPAM over SIP issues, the WG will define an token based
> authorization scheme so that domain using SIP to federate can
> choose to check that incoming SIP calls are from a domain that
> successfully validated a phone number.
Same here, you state a solution to a pb.  I would prefer to have a requirem=
ent in the charter that mandates the WG comes up with a method to help miti=
gate SPIT by ensuring that a domain using SIP can validate incoming calls a=
re indeed from a domain that successfully validated the TN.


> The problem statement and some possible starting points for
> solutions are further desired in the following internet drafts
> which shall form the bases of the WG documents:
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
>=20
> The working group will carefully coordinate with the security area,
> O&M area, as well as the appropriate RAI WG including sipcore and
> p2psip.
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From jf.mule@cablelabs.com  Wed Jun  2 14:02:16 2010
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To: Peter Musgrave <peter.musgrave@magorcorp.com>
Date: Wed, 2 Jun 2010 15:01:57 -0600
Thread-Topic: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Peter Musgrave wrote:
> Does the charter need to say anything explicitly about not relying
> on
> any of the work which uses ENUMs in DNS as a way of associating SIP
> endpoints and IP addresses?

I don't think the charter should exclude anything that may be used in some =
context as validation schemes. =20

Jean-Fran=E7ois

From peter.musgrave@magorcorp.com  Wed Jun  2 14:02:43 2010
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>Hi,
>
>  I support this proposed work to be taken by IETF under a new working group.  There are certainly concrete deliverables that would merit IETF consensus to spur multi-vendor >interoperability.


+1

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From: Jean-Francois Mule <jf.mule@cablelabs.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Peter Musgrave <peter.musgrave@magorcorp.com>
Date: Wed, 2 Jun 2010 15:04:23 -0600
Thread-Topic: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Paul wrote:
> As I understand it, the new WG is about coming up with technique(s)
> that
> work and are deployable in the real world. While in principle
> public
> ENUM could be used for this, in practice it isn't deployed and so
> isn't
> really usable.=20

To Patrick's point on the ENUM WG list on 5/19, it depends what you mean by=
 ENUM here.
There are probably a lot of SIP calls routed in networks today (and SMS-MMS=
) that use some kind of ENUM protocol implementations.

Jean-Fran=E7ois

From pkyzivat@cisco.com  Wed Jun  2 14:15:09 2010
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Jean-Francois Mule wrote:
> Paul wrote:
>> As I understand it, the new WG is about coming up with technique(s)
>> that
>> work and are deployable in the real world. While in principle
>> public
>> ENUM could be used for this, in practice it isn't deployed and so
>> isn't
>> really usable. 
> 
> To Patrick's point on the ENUM WG list on 5/19, it depends what you mean by ENUM here.
> There are probably a lot of SIP calls routed in networks today (and SMS-MMS) that use some kind of ENUM protocol implementations.

I did explicitly say *public* ENUM.
I realize other variants of ENUM are in use, but they aren't relevant to 
*this* problem statement, and I think what I said is true for *public* ENUM.

Its possible that this could result in a grab bag of techniques. 
Potentially public ENUM could be tried, and used if there is a workable 
entry, while some of the more complex techniques could be tried if there 
is no workable entry in public ENUM.

	Thanks,
	Paul

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I've improved the wording in various places and carefully removed any presu=
mptions regarding the specific solution to be delivered:

ViPR Charter Proposal (a revision)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications that
more than a billion people use on a daily basis:  E.164 phone numbers
and DNS-rooted address (such as web servers and email addresses). The
federation design of SIP is primarily designed for email-style
addresses, yet a large percentage of SIP deployments primarily use
phone numbers for identifying users. The goal of this working group is
to allow people to use SIP to federate over the the Internet while
still using phone numbers to identify the person they wish to
communicate with.

The VIPR WG will address this problem by developing one or more
peer-to-peer based approaches to determining and validating the SIP
URI(s) that are equivalent to a given phone number, without requiring
reference to a global, accurate database.

One approach to be considered will be based on a domain being able to
prove it received a particular phone call over the PSTN, based on both
sides knowing the start and stop times of that call.  This approach is
described in:

draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam

Other validation techniques (such as examining fingerprints or
watermarking of PSTN media) may be considered by the working group.

To help mitigate SPAM over SIP issues, the WG also define one or more
authorization schemes allowing a recipient to verify that an
incoming SIP call is from a source that has previously executed a
validation protocol with the recipient.

The working group will carefully coordinate with the security area,
O&M area, and the appropriate RAI WGs (including sipcore and p2psip).

Dale

From jf.mule@cablelabs.com  Wed Jun  2 15:25:31 2010
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From: Jean-Francois Mule <jf.mule@cablelabs.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 2 Jun 2010 16:25:13 -0600
Thread-Topic: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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> Jean-Francois Mule wrote:
> > Paul wrote:
> >> As I understand it, the new WG is about coming up with
> technique(s)
> >> that
> >> work and are deployable in the real world. While in principle
> >> public
> >> ENUM could be used for this, in practice it isn't deployed and
> so
> >> isn't
> >> really usable.
> >
> > To Patrick's point on the ENUM WG list on 5/19, it depends what
> you mean by ENUM here.
> > There are probably a lot of SIP calls routed in networks today
> (and SMS-MMS) that use some kind of ENUM protocol implementations.
>=20
> I did explicitly say *public* ENUM.
You're right.

> I realize other variants of ENUM are in use, but they aren't
> relevant to
> *this* problem statement, and I think what I said is true for
> *public* ENUM.
> Its possible that this could result in a grab bag of techniques.
> Potentially public ENUM could be tried, and used if there is a
> workable
> entry, while some of the more complex techniques could be tried if
> there
> is no workable entry in public ENUM.

Yes.

From alex@vidyo.com  Wed Jun  2 16:21:20 2010
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From: Alex Eleftheriadis <alex@vidyo.com>
To: =?Windows-1252?Q?H=E5kon_Dahle?= <hakon.dahle@tandberg.com>
Date: Wed, 2 Jun 2010 19:20:39 -0400
Thread-Topic: [dispatch] Updated charter/work description for telepresence multi-streams
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Subject: Re: [dispatch] Updated charter/work description for telepresence	multi-streams
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Inline.

On Jun 2, 2010, at 9:51 PM, H=E5kon Dahle wrote:

A comment on the following sentence in the updated charter:

=93This requires considering current widely deployed use cases, such as sin=
gle and multiple screens, multi-point, Scalable Video Coding (SVC), as well=
 as cases that are expected to be implemented using the protocol framework =
produced by this work item.  =94

Addressing use cases such as single and multiple screen systems, as well as=
 both point-to-point and multi-point calling is great.  However I do not un=
derstand why a specific video codec or coding technique has been added to t=
he same context.  The charter shouldn=92t recommend or limit the work for a=
ny specific codec (it should be codec agnostic),  and I suggest the charter=
 should be changed to comply with this.

Right. I believe that the correct reference is to alternative architectures=
 for the server in both point-to-point and multi-point calls, where scalabi=
lity (which includes simulcasting, pyramidal coding such as SVC, and multip=
le description coding) lends itself to VRUs (Video Routing Units) as oppose=
d to MCUs. The VRU employs a "subscribe" model, made possible by scalabilit=
y, and which works quite differently than an MCU.

--Alex



-h

________________________________
Fra: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> p=E5 vegne=
 av Allyn Romanow (allyn)
Sendt: on 02.06.2010 18:11
Til: dispatch@ietf.org<mailto:dispatch@ietf.org>; mary.ietf.barnes@gmail.co=
m<mailto:mary.ietf.barnes@gmail.com>; Cullen Jennings (fluffy)
Emne: [dispatch] Updated charter/work description for telepresencemulti-str=
eams
Folks,
Here is a version of the work description that addresses the comments made =
on the first draft.
The changes include:
1.       Charter section, qualification that we are working on SIP-based sy=
stems
2.       Charter section, minor word smithing
3.       Charter section, reference to RAI WG with relevant work
4.       Scope section, clarification of interoperability requirements
5.       Scope section, replacement of second paragraph regarding treatment=
 of non-video non-audio media types
6.       Goals and Milestones =96 addition of a Problem Statement draft


Comments welcome

Allyn

Multi-streams for Telepresence Description of Work



Background
One branch of video conferencing has evolved that is focused on immersive =
=93being there=94 experience.  Referred to in various ways such as virtual =
conferencing, telepresence or media spaces, early systems were mainly resea=
rch projects or business systems with limited deployments.  In recent years=
 telepresence systems have seen considerable market success.  Following the=
 model of early systems, the first wave of commercial systems have been typ=
ically located in specially designed single-purpose rooms with multiple rel=
atively large displays permitting life size image reproduction, multiple ca=
meras, encoders, decoders and microphones.  These systems have several impo=
rtant characteristics that are different from more traditional video confer=
encing systems.

The first difference concerns controlling the visual viewpoint in order to =
improve participant nonverbal communication. These systems preserve essenti=
al group meeting characteristics such as eye contact, group gestures, seati=
ng order and spatial audio by carefully orchestrating the miking and camera=
 angles at each of the sites . This is distinct from the more traditional a=
pproach where the geometric relationship between media streams is not used =
to preserve inter-stream communication aspects such as eye contact and grou=
p dynamics.

A second difference is manipulation of the environment to improve immersion=
.  With telepresence systems, cinematographic aspects of the local environm=
ent reproduction are carefully planned including color, table shape, seatin=
g and lighting so that when combined with large high quality displays, a st=
rong sense of a "trompe l'oeil" or "being there" immersive experience is cr=
eated.  Typical video conference systems do not include these consideration=
s.

As telepresence video systems have become successful in the market, manufac=
turers have started exploring delivery of the nonverbal communication and i=
mmersive values of telepresence via smaller, less expensive and more flexib=
le video conferencing systems for a variety of venues, such as individual o=
ffices, homes and kiosks. These are also  telepresence systems, since the a=
udio and video quality is high enough to allow clear image reproduction for=
 nonverbal communication, they are able to send and receive multiple media =
streams, and large coordinated multi image displays are available for immer=
sive installations.   As the industry develops, the line between telepresen=
ce and video conferencing may become blurred as nonverbal communication and=
 immersive installations become broadly available.

Problem
Although telepresence systems are based on open standards such as RTP, SIP,=
 H.264, H.323 suite, they cannot easily interoperate with each other withou=
t operator assistance and expensive additional equipment that translates fr=
om one vendor to another. It would be like having to make sure all parties =
are on the same equipment (and network) when making a telephone call.  A ma=
jor factor in the inability of Telepresence systems to work with each other=
 is that there is no standard description of the multiple streams that comp=
rise the media flows.

For example, in a multiple screen conference, the video and audio streams s=
ent from remote participants must be understood by receivers so that they c=
an be presented in a coherent and life-like manner. This includes the abili=
ty to present the remote participants at their true size for their apparent=
 distance, while maintaining correct eye contact, gesticular cues, and simu=
ltaneously providing a spatial audio sound stage consistent with the video =
presentation.  The receiving device that decides how to display the incomin=
g information needs to understand a number of variables such as the spatial=
 position of the speaker, the field of view of the cameras, the camera zoom=
, which media stream is related to each of the displays, etc.

Charter
The Telepresence Multi-Streams work item in DISPATCH is chartered to define=
 and specify for SIP-based systems the content of media multi-stream messag=
es and the way these will be transported.

This work will provide a standard for the exchange of media semantic inform=
ation that will foster interoperable end stations and conference bridges. I=
t will specify  variables that describe the semantics of the media streams =
and the recommended behavior to achieve interoperability.

This requires considering current widely deployed use cases, such as single=
 and multiple screens, multi-point, Scalable Video Coding (SVC), as well as=
 cases that are expected to be implemented using the protocol framework pro=
duced by this work item.  The methodology for describing the variables must=
 allow extensibility of the variables, since telepresence is still a young =
technology and may have use cases that are not currently considered.

The work item will identify use cases, define requirements, and define a me=
thod for describing and transporting information about multiple media strea=
ms, including a specification of variables required to support the use case=
s. This work item will consider the reuse of existing IETF protocols and pr=
oduce an architecture/protocol framework document describing the protocols =
required to be implemented to support this functionality.  The document wil=
l identify any enhancements required to existing protocols as well as descr=
ibing new protocol(s) for interoperable multi-streams negotiation that may =
be required.

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and FECf=
rame.

Scope
The scope includes both systems that provide a fully immersive experience, =
and systems that interwork with them and therefore need to understand the s=
ame multiple stream semantics.


The focus of this work is on audio and video multiple streams.  Other media=
 types may be considered, however development of methodologies for them is =
not within the scope of this work.

Interoperation with standards compliant systems is required, such as SIP-ba=
sed video conferencing systems.  However, backwards compatibility with exis=
ting non-standards compliant telepresence systems is not required.



The group will produce
- Requirements and use cases

- Architectural Framework describing the protocols required to be implement=
ed to support this functionality and identifying existing protocol enhancem=
ents and new protocol functionality required

- Specification of a new protocol to support telepresence multi-streams [if=
 required]

Goals and Milestones
Nov 2010


Use Cases and Requirements to IESG as Informational RFC

Nov 2010
March 2011

Problem Statement to IESG as Informational RFC
Architecture to IESG as Informational RFC

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011

Submit protocol draft to IESG as Proposed Standard RFC




_______________________________________________
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dispatch@ietf.org<mailto:dispatch@ietf.org>
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<html><head><base href=3D"x-msg://52/"></head><body style=3D"word-wrap: bre=
ak-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "=
>Inline.&nbsp;<div><br><div><div>On Jun 2, 2010, at 9:51 PM, H=E5kon Dahle =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote type=3D"cit=
e"><span class=3D"Apple-style-span" style=3D"border-collapse: separate; fon=
t-family: Helvetica; font-size: medium; font-style: normal; font-variant: n=
ormal; font-weight: normal; letter-spacing: normal; line-height: normal; or=
phans: 2; text-indent: 0px; text-transform: none; white-space: normal; wido=
ws: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-b=
order-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -web=
kit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div lang=3D"=
NO-BOK" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1"><div sty=
le=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-l=
eft: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=
=3D"EN-US" style=3D"font-size: 12pt; ">A comment on the following sentence =
in the updated charter:<o:p></o:p></span></div><div style=3D"margin-top: 0c=
m; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size:=
 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"fo=
nt-size: 12pt; "><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0c=
m; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size:=
 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"fo=
nt-size: 12pt; ">=93This requires considering current widely deployed use c=
ases, such as single and multiple screens, multi-point, Scalable Video Codi=
ng (SVC), as well as cases that are expected to be implemented using the pr=
otocol framework produced by this work item.&nbsp; =94<o:p></o:p></span></d=
iv><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001p=
t; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><=
span lang=3D"EN-US" style=3D"font-size: 12pt; "><o:p>&nbsp;</o:p></span></d=
iv><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001p=
t; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><=
span lang=3D"EN-US" style=3D"color: rgb(31, 73, 125); ">Addressing use case=
s such as single and multiple screen systems, as well as both point-to-poin=
t and multi-point calling is great. &nbsp;However I do not understand why a=
 specific video codec or coding technique has been added to the same contex=
t.&nbsp; The charter shouldn=92t recommend or limit the work for any specif=
ic codec (it should be codec agnostic),&nbsp; and I suggest the charter sho=
uld be changed to comply with this.&nbsp;<o:p></o:p></span></div><div style=
=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-lef=
t: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"=
EN-US" style=3D"font-size: 12pt; color: rgb(31, 73, 125); "></span></div></=
div></div></span></blockquote><div><br></div><div>Right. I believe that the=
 correct reference is to alternative architectures for the server in both p=
oint-to-point and multi-point calls, where scalability (which includes simu=
lcasting, pyramidal coding such as SVC, and multiple description coding) le=
nds itself to VRUs (Video Routing Units) as opposed to MCUs. The VRU employ=
s a "subscribe" model, made possible by scalability, and which works quite =
differently than an MCU.&nbsp;</div><div><br></div><div>--Alex</div><div><b=
r></div><br><blockquote type=3D"cite"><span class=3D"Apple-style-span" styl=
e=3D"border-collapse: separate; font-family: Helvetica; font-size: medium; =
font-style: normal; font-variant: normal; font-weight: normal; letter-spaci=
ng: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transfo=
rm: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border=
-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-tex=
t-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text=
-stroke-width: 0px; "><div lang=3D"NO-BOK" link=3D"blue" vlink=3D"purple"><=
div class=3D"WordSection1"><div style=3D"margin-top: 0cm; margin-right: 0cm=
; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: =
Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-size: 12pt; color=
: rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div style=3D"margin-to=
p: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-=
size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=
=3D"font-size: 12pt; font-family: 'Times New Roman', serif; color: rgb(31, =
73, 125); ">-h<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin=
-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; fo=
nt-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-size: =
12pt; font-family: 'Times New Roman', serif; color: rgb(31, 73, 125); "><o:=
p>&nbsp;</o:p></span></div><div class=3D"MsoNormal" align=3D"center" style=
=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-lef=
t: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; text-align: cent=
er; "><span lang=3D"EN-US" style=3D"font-size: 12pt; font-family: 'Times Ne=
w Roman', serif; "><hr size=3D"2" width=3D"100%" align=3D"center"></span></=
div><p class=3D"MsoNormal" style=3D"margin-top: 0cm; margin-right: 0cm; mar=
gin-bottom: 12pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, =
sans-serif; "><b><span lang=3D"EN-US" style=3D"font-size: 10pt; font-family=
: Tahoma, sans-serif; ">Fra:</span></b><span lang=3D"EN-US" style=3D"font-s=
ize: 10pt; font-family: Tahoma, sans-serif; "><span class=3D"Apple-converte=
d-space">&nbsp;</span><a href=3D"mailto:dispatch-bounces@ietf.org" style=3D=
"color: blue; text-decoration: underline; ">dispatch-bounces@ietf.org</a><s=
pan class=3D"Apple-converted-space">&nbsp;</span>p=E5 vegne av Allyn Romano=
w (allyn)<br><b>Sendt:</b><span class=3D"Apple-converted-space">&nbsp;</spa=
n>on 02.06.2010 18:11<br><b>Til:</b><span class=3D"Apple-converted-space">&=
nbsp;</span><a href=3D"mailto:dispatch@ietf.org" style=3D"color: blue; text=
-decoration: underline; ">dispatch@ietf.org</a>;<span class=3D"Apple-conver=
ted-space">&nbsp;</span><a href=3D"mailto:mary.ietf.barnes@gmail.com" style=
=3D"color: blue; text-decoration: underline; ">mary.ietf.barnes@gmail.com</=
a>; Cullen Jennings (fluffy)<br><b>Emne:</b><span class=3D"Apple-converted-=
space">&nbsp;</span>[dispatch] Updated charter/work description for telepre=
sencemulti-streams</span><span lang=3D"EN-US" style=3D"font-size: 12pt; fon=
t-family: 'Times New Roman', serif; "><o:p></o:p></span></p><div><div style=
=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-lef=
t: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"=
EN-US">Folks,<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-=
right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; fon=
t-family: Calibri, sans-serif; "><span lang=3D"EN-US">Here is a version of =
the work description that addresses the comments made on the first draft.<o=
:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; mar=
gin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calib=
ri, sans-serif; "><span lang=3D"EN-US">The changes include:<o:p></o:p></spa=
n></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.=
0001pt; margin-left: 36pt; font-size: 11pt; font-family: Calibri, sans-seri=
f; text-indent: -18pt; "><span lang=3D"EN-US">1.</span><span lang=3D"EN-US"=
 style=3D"font-size: 7pt; font-family: 'Times New Roman', serif; ">&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D"Apple-converted-space">&nbsp;</s=
pan></span><span lang=3D"EN-US">Charter section, qualification that we are =
working on SIP-based systems<o:p></o:p></span></div><div style=3D"margin-to=
p: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 36pt; font=
-size: 11pt; font-family: Calibri, sans-serif; text-indent: -18pt; "><span =
lang=3D"EN-US">2.</span><span lang=3D"EN-US" style=3D"font-size: 7pt; font-=
family: 'Times New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<sp=
an class=3D"Apple-converted-space">&nbsp;</span></span><span lang=3D"EN-US"=
>Charter section, minor word smithing<o:p></o:p></span></div><div style=3D"=
margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 3=
6pt; font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -18pt;=
 "><span lang=3D"EN-US">3.</span><span lang=3D"EN-US" style=3D"font-size: 7=
pt; font-family: 'Times New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span></span><span lang=
=3D"EN-US">Charter section, reference to RAI WG with relevant work<o:p></o:=
p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bot=
tom: 0.0001pt; margin-left: 36pt; font-size: 11pt; font-family: Calibri, sa=
ns-serif; text-indent: -18pt; "><span lang=3D"EN-US">4.</span><span lang=3D=
"EN-US" style=3D"font-size: 7pt; font-family: 'Times New Roman', serif; ">&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span class=3D"Apple-converted-space">&n=
bsp;</span></span><span lang=3D"EN-US">Scope section, clarification of inte=
roperability requirements<o:p></o:p></span></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 36pt; font-si=
ze: 11pt; font-family: Calibri, sans-serif; text-indent: -18pt; "><span lan=
g=3D"EN-US">5.</span><span lang=3D"EN-US" style=3D"font-size: 7pt; font-fam=
ily: 'Times New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span><span lang=3D"EN-US">Sc=
ope section, replacement of second paragraph regarding treatment of non-vid=
eo non-audio media types<o:p></o:p></span></div><div style=3D"margin-top: 0=
cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 36pt; font-siz=
e: 11pt; font-family: Calibri, sans-serif; text-indent: -18pt; "><span lang=
=3D"EN-US">6.</span><span lang=3D"EN-US" style=3D"font-size: 7pt; font-fami=
ly: 'Times New Roman', serif; ">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span c=
lass=3D"Apple-converted-space">&nbsp;</span></span><span lang=3D"EN-US">Goa=
ls and Milestones =96 addition of a Problem Statement draft<o:p></o:p></spa=
n></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.=
0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif=
; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin=
-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nb=
sp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm=
; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: =
Calibri, sans-serif; "><span lang=3D"EN-US">Comments welcome<o:p></o:p></sp=
an></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0=
.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-seri=
f; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margi=
n-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; f=
ont-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">Al=
lyn<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm=
; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: =
Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><=
div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; m=
argin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; text-al=
ign: center; "><b><span lang=3D"EN-US" style=3D"font-family: Arial, sans-se=
rif; ">Multi-streams for Telepresence Description of Work</span></b><span l=
ang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin=
-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; fo=
nt-family: Calibri, sans-serif; text-align: center; "><span lang=3D"EN-US">=
&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: =
0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-famil=
y: Calibri, sans-serif; text-align: center; "><span lang=3D"EN-US">&nbsp;<o=
:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; mar=
gin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calib=
ri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div s=
tyle=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin=
-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><b><span l=
ang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">Background</span><=
/b><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0=
cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size=
: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"f=
ont-family: Arial, sans-serif; ">One branch of video conferencing has evolv=
ed that is focused on immersive =93being there=94 experience.&nbsp; Referre=
d to in various ways such as virtual conferencing, telepresence or media sp=
aces, early systems were mainly research projects or business systems with =
limited deployments.&nbsp; In recent years telepresence systems have seen c=
onsiderable market success.&nbsp; Following the model of early systems, the=
 first wave of commercial systems have been typically located in specially =
designed single-purpose rooms with multiple relatively large displays permi=
tting life size image reproduction, multiple cameras, encoders, decoders an=
d microphones.&nbsp; These systems have several important characteristics t=
hat are different from more traditional video conferencing systems.&nbsp;</=
span><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top:=
 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-si=
ze: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o=
:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; mar=
gin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calib=
ri, sans-serif; "><span lang=3D"EN-US" style=3D"font-family: Arial, sans-se=
rif; ">The first difference concerns controlling the visual viewpoint in or=
der to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group gestures=
, seating order and spatial audio by carefully orchestrating the miking and=
 camera angles at each of the sites . This is distinct from the more tradit=
ional approach where the geometric relationship between media streams is no=
t used to preserve inter-stream communication aspects such as eye contact a=
nd group dynamics.&nbsp;</span><span lang=3D"EN-US"><o:p></o:p></span></div=
><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt;=
 margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><sp=
an lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0=
cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size=
: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"f=
ont-family: Arial, sans-serif; ">A second difference is manipulation of the=
 environment to improve immersion.&nbsp; With telepresence systems, cinemat=
ographic aspects of the local environment reproduction are carefully planne=
d including color, table shape, seating and lighting so that when combined =
with large high quality displays, a strong sense of a "trompe l'oeil" or "b=
eing there" immersive experience is created.&nbsp; Typical video conference=
 systems do not include these considerations.</span><span lang=3D"EN-US"><o=
:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; mar=
gin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calib=
ri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div s=
tyle=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin=
-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=
=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">As telepresence video=
 systems have become successful in the market, manufacturers have started e=
xploring delivery of the nonverbal communication and immersive values of te=
lepresence via smaller, less expensive and more flexible video conferencing=
 systems for a variety of venues, such as individual offices, homes and kio=
sks. These are also&nbsp; telepresence systems, since the audio and video q=
uality is high enough to allow clear image reproduction for nonverbal commu=
nication, they are able to send and receive multiple media streams, and lar=
ge coordinated multi image displays are available for immersive installatio=
ns.&nbsp;&nbsp; As the industry develops, the line between telepresence and=
 video conferencing may become blurred as nonverbal communication and immer=
sive installations become broadly available.</span><span lang=3D"EN-US"><o:=
p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; marg=
in-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibr=
i, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div st=
yle=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-=
left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><b><span la=
ng=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">Problem</span></b><=
span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11=
pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-=
family: Arial, sans-serif; ">Although telepresence systems are based on ope=
n standards such as RTP, SIP, H.264, H.323 suite, they cannot easily intero=
perate with each other without operator assistance and expensive additional=
 equipment that translates from one vendor to another. It would be like hav=
ing to make sure all parties are on the same equipment (and network) when m=
aking a telephone call. &nbsp;A major factor in the inability of Telepresen=
ce systems to work with each other is that there is no standard description=
 of the multiple streams that comprise the media flows.</span><span lang=3D=
"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right=
: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-fam=
ily: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></=
div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001=
pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; ">=
<span lang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">For example=
, in a multiple screen conference, the video and audio streams sent from re=
mote participants must be understood by receivers so that they can be prese=
nted in a coherent and life-like manner. This includes the ability to prese=
nt the remote participants at their true size for their apparent distance, =
while maintaining correct eye contact, gesticular cues, and simultaneously =
providing a spatial audio sound stage consistent with the video presentatio=
n.&nbsp; The receiving device that decides how to display the incoming info=
rmation needs to understand a number of variables such as the spatial posit=
ion of the speaker, the field of view of the cameras, the camera zoom, whic=
h media stream is related to each of the displays, etc.</span><span lang=3D=
"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right=
: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-fam=
ily: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></=
div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001=
pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; ">=
<b><span lang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">Charter<=
/span></b><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin=
-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" sty=
le=3D"font-family: Arial, sans-serif; ">The Telepresence Multi-Streams work=
 item in DISPATCH is chartered to define and specify for SIP-based systems =
the content of media multi-stream messages and the way these will be transp=
orted.</span><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"mar=
gin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm;=
 font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">=
&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: =
0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-famil=
y: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-family: Arial,=
 sans-serif; ">This work will provide a standard for the exchange of media =
semantic information that will foster interoperable end stations and confer=
ence bridges. It will specify&nbsp; variables that describe the semantics o=
f the media streams and the recommended behavior to achieve interoperabilit=
y.&nbsp;</span><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"m=
argin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0c=
m; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US=
">&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right=
: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-fam=
ily: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-family: Aria=
l, sans-serif; ">This requires considering current widely deployed use case=
s, such as single and multiple screens, multi-point, Scalable Video Coding =
(SVC), as well as cases that are expected to be implemented using the proto=
col framework produced by this work item.&nbsp; The methodology for describ=
ing the variables must allow extensibility of the variables, since telepres=
ence is still a young technology and may have use cases that are not curren=
tly considered.</span><span lang=3D"EN-US"><o:p></o:p></span></div><div sty=
le=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-l=
eft: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=
=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; mar=
gin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt;=
 font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-fam=
ily: Arial, sans-serif; ">The work item will identify use cases, define req=
uirements, and define a method for describing and transporting information =
about multiple media streams, including a specification of variables requir=
ed to support the use cases. This work item will consider the reuse of exis=
ting IETF protocols and produce an architecture/protocol framework document=
 describing the protocols required to be implemented to support this functi=
onality.&nbsp; The document will identify any enhancements required to exis=
ting protocols as well as describing new protocol(s) for interoperable mult=
i-streams negotiation that may be required.</span><span lang=3D"EN-US"><o:p=
></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margi=
n-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri=
, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div sty=
le=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-l=
eft: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=
=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">Relevant work to be d=
rawn upon has been done in XCON, MMUSIC, AVT, and FECframe.</span><span lan=
g=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-r=
ight: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font=
-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></spa=
n></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.=
0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif=
; "><b><span lang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">Scop=
e</span></b><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"marg=
in-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; =
font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" s=
tyle=3D"font-family: Arial, sans-serif; ">The scope includes both systems t=
hat provide a fully immersive experience, and systems that interwork with t=
hem and therefore need to understand the same multiple stream semantics.</s=
pan><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-siz=
e: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:=
p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; marg=
in-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibr=
i, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div st=
yle=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-=
left: 0cm; font-size: 10.5pt; font-family: Consolas; "><span lang=3D"EN-US"=
 style=3D"font-size: 12pt; font-family: Arial, sans-serif; ">The focus of t=
his work is on audio and video multiple streams.&nbsp; Other media types ma=
y be considered, however development of methodologies for them is not withi=
n the scope of this work.</span><span lang=3D"EN-US"><o:p></o:p></span></di=
v><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt=
; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><s=
pan lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-siz=
e: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"=
font-family: Arial, sans-serif; ">Interoperation with standards compliant s=
ystems is required, such as SIP-based video conferencing systems. &nbsp;How=
ever, backwards compatibility with existing non-standards compliant telepre=
sence systems is not required.</span><span lang=3D"EN-US"><o:p></o:p></span=
></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0=
001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif;=
 "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin-=
top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fon=
t-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbs=
p;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm;=
 margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: C=
alibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div><d=
iv style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; ma=
rgin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><b><sp=
an lang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">The group will=
 produce</span></b><span lang=3D"EN-US"><o:p></o:p></span></div><div style=
=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-lef=
t: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"=
EN-US" style=3D"font-family: Arial, sans-serif; ">- Requirements and use ca=
ses</span><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin=
-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nb=
sp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm=
; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: =
Calibri, sans-serif; "><span lang=3D"EN-US" style=3D"font-family: Arial, sa=
ns-serif; ">- Architectural Framework describing the protocols required to =
be implemented to support this functionality and identifying existing proto=
col enhancements and new protocol functionality required</span><span lang=
=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-ri=
ght: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-=
family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span=
></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0=
001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif;=
 "><span lang=3D"EN-US" style=3D"font-family: Arial, sans-serif; ">- Specif=
ication of a new protocol to support telepresence multi-streams [if require=
d]</span><span lang=3D"EN-US"><o:p></o:p></span></div><div style=3D"margin-=
top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fon=
t-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbs=
p;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm;=
 margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: C=
alibri, sans-serif; "><b><span lang=3D"EN-US" style=3D"font-family: Arial, =
sans-serif; ">Goals and Milestones</span></b><span lang=3D"EN-US"><o:p></o:=
p></span></div><table class=3D"MsoNormalTable" border=3D"0" cellpadding=3D"=
0"><tbody><tr><td width=3D"96" style=3D"width: 72pt; padding-top: 0.75pt; p=
adding-right: 0.75pt; padding-bottom: 0.75pt; padding-left: 0.75pt; "><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margi=
n-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span sty=
le=3D"font-family: Arial, sans-serif; ">Nov 2010</span><o:p></o:p></div></t=
d><td width=3D"381" style=3D"width: 285.75pt; padding-top: 0.75pt; padding-=
right: 0.75pt; padding-bottom: 0.75pt; padding-left: 0.75pt; "><div style=
=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-lef=
t: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; ">&nbsp;<o:p></o=
:p></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0=
.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-seri=
f; "><span style=3D"font-family: Arial, sans-serif; ">Use Cases and Require=
ments to IESG as Informational RFC</span><o:p></o:p></div></td></tr><tr><td=
 width=3D"96" style=3D"width: 72pt; padding-top: 0.75pt; padding-right: 0.7=
5pt; padding-bottom: 0.75pt; padding-left: 0.75pt; "><div style=3D"margin-t=
op: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font=
-size: 11pt; font-family: Calibri, sans-serif; "><span style=3D"font-family=
: Arial, sans-serif; ">Nov 2010</span><o:p></o:p></div><div style=3D"margin=
-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; fo=
nt-size: 11pt; font-family: Calibri, sans-serif; "><span style=3D"font-fami=
ly: Arial, sans-serif; ">March 2011</span><o:p></o:p></div></td><td width=
=3D"381" style=3D"width: 285.75pt; padding-top: 0.75pt; padding-right: 0.75=
pt; padding-bottom: 0.75pt; padding-left: 0.75pt; "><div style=3D"margin-to=
p: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-=
size: 11pt; font-family: Calibri, sans-serif; "><span style=3D"font-family:=
 Arial, sans-serif; ">Problem Statement to IESG as Informational RFC</span>=
<o:p></o:p></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-b=
ottom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, s=
ans-serif; "><span style=3D"font-family: Arial, sans-serif; ">Architecture =
to IESG as Informational RFC</span><o:p></o:p></div></td></tr><tr><td width=
=3D"96" style=3D"width: 72pt; padding-top: 0.75pt; padding-right: 0.75pt; p=
adding-bottom: 0.75pt; padding-left: 0.75pt; "><div style=3D"margin-top: 0c=
m; margin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size:=
 11pt; font-family: Calibri, sans-serif; "><span style=3D"font-family: Aria=
l, sans-serif; ">March 2011</span><o:p></o:p></div></td><td width=3D"381" s=
tyle=3D"width: 285.75pt; padding-top: 0.75pt; padding-right: 0.75pt; paddin=
g-bottom: 0.75pt; padding-left: 0.75pt; "><div style=3D"margin-top: 0cm; ma=
rgin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11pt=
; font-family: Calibri, sans-serif; "><span style=3D"font-family: Arial, sa=
ns-serif; ">Revise Charter [IF new protocol is not required]</span><o:p></o=
:p></div></td></tr><tr><td width=3D"96" style=3D"width: 72pt; padding-top: =
0.75pt; padding-right: 0.75pt; padding-bottom: 0.75pt; padding-left: 0.75pt=
; "><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001=
pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; ">=
<span style=3D"font-family: Arial, sans-serif; ">Nov 2011</span><o:p></o:p>=
</div></td><td width=3D"381" style=3D"width: 285.75pt; padding-top: 0.75pt;=
 padding-right: 0.75pt; padding-bottom: 0.75pt; padding-left: 0.75pt; "><di=
v style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; mar=
gin-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span s=
tyle=3D"font-family: Arial, sans-serif; ">Submit protocol draft to IESG as =
Proposed Standard RFC</span><o:p></o:p></div></td></tr></tbody></table><div=
 style=3D"margin-top: 0cm; margin-right: 0cm; margin-bottom: 0.0001pt; marg=
in-left: 0cm; font-size: 11pt; font-family: Calibri, sans-serif; "><span la=
ng=3D"EN-US">&nbsp;<o:p></o:p></span></div><div style=3D"margin-top: 0cm; m=
argin-right: 0cm; margin-bottom: 0.0001pt; margin-left: 0cm; font-size: 11p=
t; font-family: Calibri, sans-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:=
p></span></div><div style=3D"margin-top: 0cm; margin-right: 0cm; margin-bot=
tom: 0.0001pt; margin-left: 0cm; font-size: 11pt; font-family: Calibri, san=
s-serif; "><span lang=3D"EN-US">&nbsp;<o:p></o:p></span></div></div></div>_=
______________________________________________<br>dispatch mailing list<br>=
<a href=3D"mailto:dispatch@ietf.org" style=3D"color: blue; text-decoration:=
 underline; ">dispatch@ietf.org</a><br><a href=3D"https://www.ietf.org/mail=
man/listinfo/dispatch" style=3D"color: blue; text-decoration: underline; ">=
https://www.ietf.org/mailman/listinfo/dispatch</a><br></div></span></blockq=
uote></div><br></div></body></html>=

--_000_C8B9CF05B89B4946A14AA1F6FABDAE4Cvidyocom_--

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Agree with Alex to a large extent in that  scalability, simulcast, =
publish/subscribe may all be applicable to the problem at hand.=0A=
=0A=
However I do not think the Charter should jump to conclusions on what =
happens inside "the server" that Alex is referring to in his email - at =
this point I suggest leaving these details out of the charter, to make =
sure all options can be investigated.=0A=
=0A=
-h=0A=
=0A=
=0A=
=0A=
----- Reply message -----=0A=
From: "Alex Eleftheriadis" <alex@vidyo.com>=0A=
Date: Wed, Jun 2, 2010 16:20=0A=
Subject: [dispatch] Updated charter/work description for =
telepresencemulti-streams=0A=
To: "H=C3=A5kon Dahle" <hakon.dahle@tandberg.com>=0A=
Cc: "dispatch@ietf.org" <dispatch@ietf.org>=0A=

Inline.

On Jun 2, 2010, at 9:51 PM, H=C3=A5kon Dahle wrote:

A comment on the following sentence in the updated charter:

=E2=80=9CThis requires considering current widely deployed use cases, =
such as single and multiple screens, multi-point, Scalable Video Coding =
(SVC), as well as cases that are expected to be implemented using the =
protocol framework produced by this work item.  =E2=80=9D

Addressing use cases such as single and multiple screen systems, as well =
as both point-to-point and multi-point calling is great.  However I do =
not understand why a specific video codec or coding technique has been =
added to the same context.  The charter shouldn=E2=80=99t recommend or =
limit the work for any specific codec (it should be codec agnostic),  =
and I suggest the charter should be changed to comply with this.

Right. I believe that the correct reference is to alternative =
architectures for the server in both point-to-point and multi-point =
calls, where scalability (which includes simulcasting, pyramidal coding =
such as SVC, and multiple description coding) lends itself to VRUs =
(Video Routing Units) as opposed to MCUs. The VRU employs a "subscribe" =
model, made possible by scalability, and which works quite differently =
than an MCU.

--Alex



-h

________________________________
Fra: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> p=C3=A5 =
vegne av Allyn Romanow (allyn)
Sendt: on 02.06.2010 18:11
Til: dispatch@ietf.org<mailto:dispatch@ietf.org>; =
mary.ietf.barnes@gmail.com<mailto:mary.ietf.barnes@gmail.com>; Cullen =
Jennings (fluffy)
Emne: [dispatch] Updated charter/work description for =
telepresencemulti-streams
Folks,
Here is a version of the work description that addresses the comments =
made on the first draft.
The changes include:
1.       Charter section, qualification that we are working on SIP-based =
systems
2.       Charter section, minor word smithing
3.       Charter section, reference to RAI WG with relevant work
4.       Scope section, clarification of interoperability requirements
5.       Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types
6.       Goals and Milestones =E2=80=93 addition of a Problem Statement =
draft


Comments welcome

Allyn

Multi-streams for Telepresence Description of Work



Background
One branch of video conferencing has evolved that is focused on =
immersive =E2=80=9Cbeing there=E2=80=9D experience.  Referred to in =
various ways such as virtual conferencing, telepresence or media spaces, =
early systems were mainly research projects or business systems with =
limited deployments.  In recent years telepresence systems have seen =
considerable market success.  Following the model of early systems, the =
first wave of commercial systems have been typically located in =
specially designed single-purpose rooms with multiple relatively large =
displays permitting life size image reproduction, multiple cameras, =
encoders, decoders and microphones.  These systems have several =
important characteristics that are different from more traditional video =
conferencing systems.

The first difference concerns controlling the visual viewpoint in order =
to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group =
gestures, seating order and spatial audio by carefully orchestrating the =
miking and camera angles at each of the sites . This is distinct from =
the more traditional approach where the geometric relationship between =
media streams is not used to preserve inter-stream communication aspects =
such as eye contact and group dynamics.

A second difference is manipulation of the environment to improve =
immersion.  With telepresence systems, cinematographic aspects of the =
local environment reproduction are carefully planned including color, =
table shape, seating and lighting so that when combined with large high =
quality displays, a strong sense of a "trompe l'oeil" or "being there" =
immersive experience is created.  Typical video conference systems do =
not include these considerations.

As telepresence video systems have become successful in the market, =
manufacturers have started exploring delivery of the nonverbal =
communication and immersive values of telepresence via smaller, less =
expensive and more flexible video conferencing systems for a variety of =
venues, such as individual offices, homes and kiosks. These are also  =
telepresence systems, since the audio and video quality is high enough =
to allow clear image reproduction for nonverbal communication, they are =
able to send and receive multiple media streams, and large coordinated =
multi image displays are available for immersive installations.   As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.

Problem
Although telepresence systems are based on open standards such as RTP, =
SIP, H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call.  A major factor in the inability of Telepresence systems =
to work with each other is that there is no standard description of the =
multiple streams that comprise the media flows.

For example, in a multiple screen conference, the video and audio =
streams sent from remote participants must be understood by receivers so =
that they can be presented in a coherent and life-like manner. This =
includes the ability to present the remote participants at their true =
size for their apparent distance, while maintaining correct eye contact, =
gesticular cues, and simultaneously providing a spatial audio sound =
stage consistent with the video presentation.  The receiving device that =
decides how to display the incoming information needs to understand a =
number of variables such as the spatial position of the speaker, the =
field of view of the cameras, the camera zoom, which media stream is =
related to each of the displays, etc.

Charter
The Telepresence Multi-Streams work item in DISPATCH is chartered to =
define and specify for SIP-based systems the content of media =
multi-stream messages and the way these will be transported.

This work will provide a standard for the exchange of media semantic =
information that will foster interoperable end stations and conference =
bridges. It will specify  variables that describe the semantics of the =
media streams and the recommended behavior to achieve interoperability.

This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  The methodology for describing =
the variables must allow extensibility of the variables, since =
telepresence is still a young technology and may have use cases that are =
not currently considered.

The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media =
streams, including a specification of variables required to support the =
use cases. This work item will consider the reuse of existing IETF =
protocols and produce an architecture/protocol framework document =
describing the protocols required to be implemented to support this =
functionality.  The document will identify any enhancements required to =
existing protocols as well as describing new protocol(s) for =
interoperable multi-streams negotiation that may be required.

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.

Scope
The scope includes both systems that provide a fully immersive =
experience, and systems that interwork with them and therefore need to =
understand the same multiple stream semantics.


The focus of this work is on audio and video multiple streams.  Other =
media types may be considered, however development of methodologies for =
them is not within the scope of this work.

Interoperation with standards compliant systems is required, such as =
SIP-based video conferencing systems.  However, backwards compatibility =
with existing non-standards compliant telepresence systems is not =
required.



The group will produce
- Requirements and use cases

- Architectural Framework describing the protocols required to be =
implemented to support this functionality and identifying existing =
protocol enhancements and new protocol functionality required

- Specification of a new protocol to support telepresence multi-streams =
[if required]

Goals and Milestones
Nov 2010


Use Cases and Requirements to IESG as Informational RFC

Nov 2010
March 2011

Problem Statement to IESG as Informational RFC
Architecture to IESG as Informational RFC

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011

Submit protocol draft to IESG as Proposed Standard RFC




_______________________________________________
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Date: Wed, 2 Jun 2010 17:53:34 -0700
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Thread-Topic: [dispatch] Updated charter/work description fortelepresencemulti-streams
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From: "Allyn Romanow (allyn)" <allyn@cisco.com>
To: =?utf-8?B?SMOla29uIERhaGxl?= <hakon.dahle@tandberg.com>, <dispatch@ietf.org>, <alex@vidyo.com>
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Subject: Re: [dispatch] Updated charter/work description fortelepresencemulti-streams
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Allyn,=0A=
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I agree with your suggestion.=0A=
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=0A=
----- Reply message -----=0A=
From: "Allyn Romanow (allyn)" <allyn@cisco.com>=0A=
Date: Wed, Jun 2, 2010 17:53=0A=
Subject: [dispatch] Updated charter/work description =
fortelepresencemulti-streams=0A=
To: "H=C3=A5kon Dahle" <hakon.dahle@tandberg.com>, "dispatch@ietf.org" =
<dispatch@ietf.org>, "alex@vidyo.com" <alex@vidyo.com>=0A=

Hi Alex and Hakon,=20

I think the intent here is to say that we want to cover current use =
cases and be extensible enough to cover new ones that may arise. I don't =
think it valuable to exhaustively enumerate relevant dimensions of use =
cases in this charter statement - that is better left to the use case =
document.=20

How would it be to say,

"This requires considering current widely deployed use cases, as well as =
cases that are expected to be implemented using the protocol framework =
produced by this work item.  The methodology for describing the =
variables must allow extensibility of the variables, since telepresence =
is still a young technology and may have use cases that are not =
currently considered."


Allyn

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of H=C3=A5kon Dahle
Sent: Wednesday, June 02, 2010 5:02 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] Updated charter/work description =
fortelepresencemulti-streams


Agree with Alex to a large extent in that  scalability, simulcast, =
publish/subscribe may all be applicable to the problem at hand.

However I do not think the Charter should jump to conclusions on what =
happens inside "the server" that Alex is referring to in his email - at =
this point I suggest leaving these details out of the charter, to make =
sure all options can be investigated.

-h



----- Reply message -----
From: "Alex Eleftheriadis" <alex@vidyo.com>
Date: Wed, Jun 2, 2010 16:20
Subject: [dispatch] Updated charter/work description for =
telepresencemulti-streams
To: "H=C3=A5kon Dahle" <hakon.dahle@tandberg.com>
Cc: "dispatch@ietf.org" <dispatch@ietf.org>

Inline.

On Jun 2, 2010, at 9:51 PM, H=C3=A5kon Dahle wrote:

A comment on the following sentence in the updated charter:

=E2=80=9CThis requires considering current widely deployed use cases, =
such as single and multiple screens, multi-point, Scalable Video Coding =
(SVC), as well as cases that are expected to be implemented using the =
protocol framework produced by this work item.  =E2=80=9D

Addressing use cases such as single and multiple screen systems, as well =
as both point-to-point and multi-point calling is great.  However I do =
not understand why a specific video codec or coding technique has been =
added to the same context.  The charter shouldn=E2=80=99t recommend or =
limit the work for any specific codec (it should be codec agnostic),  =
and I suggest the charter should be changed to comply with this.

Right. I believe that the correct reference is to alternative =
architectures for the server in both point-to-point and multi-point =
calls, where scalability (which includes simulcasting, pyramidal coding =
such as SVC, and multiple description coding) lends itself to VRUs =
(Video Routing Units) as opposed to MCUs. The VRU employs a "subscribe" =
model, made possible by scalability, and which works quite differently =
than an MCU.

--Alex



-h

________________________________
Fra: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> p=C3=A5 =
vegne av Allyn Romanow (allyn)
Sendt: on 02.06.2010 18:11
Til: dispatch@ietf.org<mailto:dispatch@ietf.org>; =
mary.ietf.barnes@gmail.com<mailto:mary.ietf.barnes@gmail.com>; Cullen =
Jennings (fluffy)
Emne: [dispatch] Updated charter/work description for =
telepresencemulti-streams
Folks,
Here is a version of the work description that addresses the comments =
made on the first draft.
The changes include:
1.       Charter section, qualification that we are working on SIP-based =
systems
2.       Charter section, minor word smithing
3.       Charter section, reference to RAI WG with relevant work
4.       Scope section, clarification of interoperability requirements
5.       Scope section, replacement of second paragraph regarding =
treatment of non-video non-audio media types
6.       Goals and Milestones =E2=80=93 addition of a Problem Statement =
draft


Comments welcome

Allyn

Multi-streams for Telepresence Description of Work



Background
One branch of video conferencing has evolved that is focused on =
immersive =E2=80=9Cbeing there=E2=80=9D experience.  Referred to in =
various ways such as virtual conferencing, telepresence or media spaces, =
early systems were mainly research projects or business systems with =
limited deployments.  In recent years telepresence systems have seen =
considerable market success.  Following the model of early systems, the =
first wave of commercial systems have been typically located in =
specially designed single-purpose rooms with multiple relatively large =
displays permitting life size image reproduction, multiple cameras, =
encoders, decoders and microphones.  These systems have several =
important characteristics that are different from more traditional video =
conferencing systems.

The first difference concerns controlling the visual viewpoint in order =
to improve participant nonverbal communication. These systems preserve =
essential group meeting characteristics such as eye contact, group =
gestures, seating order and spatial audio by carefully orchestrating the =
miking and camera angles at each of the sites . This is distinct from =
the more traditional approach where the geometric relationship between =
media streams is not used to preserve inter-stream communication aspects =
such as eye contact and group dynamics.

A second difference is manipulation of the environment to improve =
immersion.  With telepresence systems, cinematographic aspects of the =
local environment reproduction are carefully planned including color, =
table shape, seating and lighting so that when combined with large high =
quality displays, a strong sense of a "trompe l'oeil" or "being there" =
immersive experience is created.  Typical video conference systems do =
not include these considerations.

As telepresence video systems have become successful in the market, =
manufacturers have started exploring delivery of the nonverbal =
communication and immersive values of telepresence via smaller, less =
expensive and more flexible video conferencing systems for a variety of =
venues, such as individual offices, homes and kiosks. These are also  =
telepresence systems, since the audio and video quality is high enough =
to allow clear image reproduction for nonverbal communication, they are =
able to send and receive multiple media streams, and large coordinated =
multi image displays are available for immersive installations.   As the =
industry develops, the line between telepresence and video conferencing =
may become blurred as nonverbal communication and immersive =
installations become broadly available.

Problem
Although telepresence systems are based on open standards such as RTP, =
SIP, H.264, H.323 suite, they cannot easily interoperate with each other =
without operator assistance and expensive additional equipment that =
translates from one vendor to another. It would be like having to make =
sure all parties are on the same equipment (and network) when making a =
telephone call.  A major factor in the inability of Telepresence systems =
to work with each other is that there is no standard description of the =
multiple streams that comprise the media flows.

For example, in a multiple screen conference, the video and audio =
streams sent from remote participants must be understood by receivers so =
that they can be presented in a coherent and life-like manner. This =
includes the ability to present the remote participants at their true =
size for their apparent distance, while maintaining correct eye contact, =
gesticular cues, and simultaneously providing a spatial audio sound =
stage consistent with the video presentation.  The receiving device that =
decides how to display the incoming information needs to understand a =
number of variables such as the spatial position of the speaker, the =
field of view of the cameras, the camera zoom, which media stream is =
related to each of the displays, etc.

Charter
The Telepresence Multi-Streams work item in DISPATCH is chartered to =
define and specify for SIP-based systems the content of media =
multi-stream messages and the way these will be transported.

This work will provide a standard for the exchange of media semantic =
information that will foster interoperable end stations and conference =
bridges. It will specify  variables that describe the semantics of the =
media streams and the recommended behavior to achieve interoperability.

This requires considering current widely deployed use cases, such as =
single and multiple screens, multi-point, Scalable Video Coding (SVC), =
as well as cases that are expected to be implemented using the protocol =
framework produced by this work item.  The methodology for describing =
the variables must allow extensibility of the variables, since =
telepresence is still a young technology and may have use cases that are =
not currently considered.

The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media =
streams, including a specification of variables required to support the =
use cases. This work item will consider the reuse of existing IETF =
protocols and produce an architecture/protocol framework document =
describing the protocols required to be implemented to support this =
functionality.  The document will identify any enhancements required to =
existing protocols as well as describing new protocol(s) for =
interoperable multi-streams negotiation that may be required.

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.

Scope
The scope includes both systems that provide a fully immersive =
experience, and systems that interwork with them and therefore need to =
understand the same multiple stream semantics.


The focus of this work is on audio and video multiple streams.  Other =
media types may be considered, however development of methodologies for =
them is not within the scope of this work.

Interoperation with standards compliant systems is required, such as =
SIP-based video conferencing systems.  However, backwards compatibility =
with existing non-standards compliant telepresence systems is not =
required.



The group will produce
- Requirements and use cases

- Architectural Framework describing the protocols required to be =
implemented to support this functionality and identifying existing =
protocol enhancements and new protocol functionality required

- Specification of a new protocol to support telepresence multi-streams =
[if required]

Goals and Milestones
Nov 2010


Use Cases and Requirements to IESG as Informational RFC

Nov 2010
March 2011

Problem Statement to IESG as Informational RFC
Architecture to IESG as Informational RFC

March 2011

Revise Charter [IF new protocol is not required]

Nov 2011

Submit protocol draft to IESG as Proposed Standard RFC




_______________________________________________
dispatch mailing list
dispatch@ietf.org<mailto:dispatch@ietf.org>
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_______________________________________________
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dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From fluffy@cisco.com  Thu Jun  3 09:27:01 2010
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From: Cullen Jennings <fluffy@cisco.com>
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To: Milan Patel <Milan.Patel@InterDigital.com>, DISPATCH list <dispatch@ietf.org>
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Subject: Re: [dispatch] Charter proposal for drafts related to the tel URI
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I did a bit more thinking about how to make progress on this charter and =
have a few ideas how to get the conversation going. The currently =
charter is pretty unconstrained and does not explain what the problem is =
the WG will solve.  It roughly paraphrases to there are some bits in =
some other telephone protocols that are not yet in SIP therefore we need =
to put them in the tel URI. The IPTEL WG had pretty clear consigns that =
was not an appropriate way to approach the problem and I would be =
surprised to see that consensus has radically changed. I do think people =
would be very favorable in putting in extensions that made sense to do =
this way. My suggestion is to figure out what is the problems we are =
actually trying to solve, then find a way to solve theses problems. I do =
think there are problems to be solved that we do need some extensions - =
it's just not clear where or how to do them. To help poke at what the =
problem is, let me ask some questions that might get the conversation =
going.=20

So the information being transported is roughly A tel URI to indicate =
how the CIC code was selected. Why does the receiver need this =
information and what are the use case for it? If we can be clear about =
that, I think ti will help define what information needs to be moved and =
then we can figure out the appropriate place to put it in the protocol.=20=


In the prison use case, the use case seems to be "Allow a user to block =
phone call that come from a prison". Is that about it or is there more?

I don't understand the payphone use case. Is it "charge more the call if =
it came from a payphone?"

Can you list out all the information that is needed from the OLI/CPC and =
they we can try and sort out the use cases for what parts a SIP device =
needs to understand and what parts only need to be transported across =
the SIP network.=20

Cullen


On May 28, 2010, at 7:13 AM, Patel, Milan wrote:

> Hello,
> =20
> Attached is a charter proposal for a WG to be formed for work =
pertaining to extensions to the tel URI =96 specifically, parameters for =
carrying the DAI, CPC, OLI and RN required for primarily interworking =
scenarios to ISUP/PSTN.
> Feedback is appreciated.
> Regards,
> Milan
> =20
> =20
> From: Patel, Milan=20
> Sent: Tuesday, May 25, 2010 11:16 AM
> To: Cullen Jennings; 'mary.ietf.barnes@gmail.com'; Robert Sparks; =
'Gonzalo Camarillo'
> Cc: R.Jesske@telekom.de; 'Sumanth Channabasappa'
> Subject: Charter proposal
> =20
> Hi all,
> =20
> Attached is a charter proposal for a WG to be formed to handle =
Internet Drafts pertaining to the Tel URI. Specifically these are the =
DAI draft and the CPC/OLI draft and some further work on the RN =
parameter.
> Feedback is welcome.
> =20
> Kind regards,
> Milan
> <XXXX WG charter proposal.txt>


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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I'd like to state that I am strongly in favor of addressing this in the IET=
F. Some of this information is very important to Operator Services and Dire=
ctory Assistance, and continues to be important as we move forward towards =
implementing these with SIP. We wrote a couple of drafts awhile back about =
the need to signal certain information in SIP - e.g., draft-haluska-dispatc=
h-csi-needed-00.

As has been mentioned, this issue has been raised at least three times in t=
he past decade. I believe this is an indication that this is a problem in n=
eed of attention.







[dispatch] Charter proposal for drafts related to the tel URI
________________________________

 *   From: "Patel, Milan" <Milan.Patel at InterDigital.com<mailto:Milan.Pat=
el@DOMAIN.HIDDEN>>
 *   To: <dispatch at ietf.org<mailto:dispatch@DOMAIN.HIDDEN>>, <mary.ietf.=
barnes at gmail.com<mailto:mary.ietf.barnes@DOMAIN.HIDDEN>>, "Cullen Jennin=
gs" <fluffy at cisco.com<mailto:fluffy@DOMAIN.HIDDEN>>
 *   Date: Fri, 28 May 2010 09:13:05 -0400

________________________________
Hello,

Attached is a charter proposal for a WG to be formed for work pertaining to=
 extensions to the tel URI - specifically, parameters for carrying the DAI,=
 CPC, OLI and RN required for primarily interworking scenarios to ISUP/PSTN=
.
Feedback is appreciated.
Regards,
Milan





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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'color:black'>I&#8217;d like to state th=
at I am
strongly in favor of addressing this in the IETF. Some of this information =
is
very important to Operator Services and Directory Assistance, and continues=
 to
be important as we move forward towards implementing these with SIP. We wro=
te a
couple of drafts awhile back about the need to signal certain information i=
n
SIP - e.g., draft-haluska-dispatch-csi-needed-00. <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:black'>&nbsp;<o:p></o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:black'>As has been mentioned, thi=
s issue
has been raised at least three times in the past decade. I believe this is =
an
indication that this is a problem in need of attention. <o:p></o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:black'><o:p>&nbsp;</o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:black'><o:p>&nbsp;</o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:black'><o:p>&nbsp;</o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:black'><o:p>&nbsp;</o:p></span></=
p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<h1>[dispatch] Charter proposal for drafts related to the tel URI<o:p></o:p=
></h1>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>

<ul type=3Ddisc>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><em><span style=3D'font-family:"Calibri","san=
s-serif"'>From</span></em>:
     &quot;Patel, Milan&quot; &lt;<a href=3D"mailto:Milan.Patel@DOMAIN.HIDD=
EN">Milan.Patel
     at InterDigital.com</a>&gt;<o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><em><span style=3D'font-family:"Calibri","san=
s-serif"'>To</span></em>:
     &lt;<a href=3D"mailto:dispatch@DOMAIN.HIDDEN">dispatch at ietf.org</a>=
&gt;,
     &lt;<a href=3D"mailto:mary.ietf.barnes@DOMAIN.HIDDEN">mary.ietf.barnes=
 at
     gmail.com</a>&gt;, &quot;Cullen Jennings&quot; &lt;<a
     href=3D"mailto:fluffy@DOMAIN.HIDDEN">fluffy at cisco.com</a>&gt;<o:p><=
/o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><em><span style=3D'font-family:"Calibri","san=
s-serif"'>Date</span></em>:
     Fri, 28 May 2010 09:13:05 -0400<o:p></o:p></li>
</ul>

</span>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0 width=3D"100%"
 style=3D'width:100.0%'>
 <tr>
  <td style=3D'padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>He=
llo,</span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>&n=
bsp;</span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>At=
tached
  is a charter proposal for a WG to be formed for work pertaining to extens=
ions
  to the tel URI &#8211; specifically, parameters for carrying the DAI, CPC=
,
  OLI and RN required for primarily interworking scenarios to ISUP/PSTN. </=
span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Fe=
edback
  is appreciated. </span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Re=
gards,</span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>Mi=
lan</span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>&n=
bsp;</span><o:p></o:p></p>
  <p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto'><span
  style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>&n=
bsp;</span><span
  style=3D'font-size:12.0pt'><o:p></o:p></span></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

--_000_8B6A9EC265011E4CB70F99C64426E8C2058636D1C9rrcdteexmb2dt_--

From dworley@avaya.com  Thu Jun  3 18:30:03 2010
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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: Richard Shockey <richard@shockey.us>, 'Paul Kyzivat' <pkyzivat@cisco.com>,  "'Patel, Milan'" <Milan.Patel@InterDigital.com>
Date: Thu, 3 Jun 2010 21:25:12 -0400
Thread-Topic: [dispatch] Charter proposal for drafts related to the tel URI
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Subject: Re: [dispatch] Charter proposal for drafts related to the tel URI
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________________________________________
From: Richard Shockey [richard@shockey.us]

Though the larger issue of how to carry these types of messages or metadata
(CNAM,CIC,SPID) in SIP has not been fully addressed either.

Needless to say we have had a vigorous discussion over this ongoing issue i=
n
both DRINKS and the E2MD BOF ( NGN ENUM ) list.
________________________________________

I suspect that any truly useful solution will be similar to what is being s=
olidified regarding translation Q.850 "response codes" into SIP so that SIP=
 can be used as a transit network between two SS7 networks:  (1) As much as=
 possible, the *meaning* of the Q.850 code is translated into a SIP respons=
e code, and (2) the original Q.850 code is carried directly in a way for wh=
ich SIP is transparent, in order to be resurrected at a SIP-to-SS7 gateway.

In the case of CIC and friends, carrying the codes directly in a suitable h=
eader is quite appropriate.  But I think that the approach of attaching the=
m directly to the originating or destination URI is not architecturally sou=
nd.

Dale

From Milan.Patel@InterDigital.com  Fri Jun  4 03:41:10 2010
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Thread-Topic: [dispatch] Charter proposal for drafts related to the tel URI
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From: "Patel, Milan" <Milan.Patel@InterDigital.com>
To: "WORLEY, Dale R (Dale)" <dworley@avaya.com>, "Richard Shockey" <richard@shockey.us>, "Paul Kyzivat" <pkyzivat@cisco.com>, "Cullen Jennings" <fluffy@cisco.com>
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Subject: Re: [dispatch] Charter proposal for drafts related to the tel URI
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Hi Cullen,

One thing needs to be made clear and that is the called party isn't
required to handle the information. Especially in SIP, the CPC for
example would be subject to trust domain rules. The information is
required for SIP intermediaries, or in the case of interworking to the
PSTN, the information is required for processing within the PSTN and not
by the endpoint device or user.=20

Dale, the solutions currently proposed for the CPC/OLI/DAI is something
that I believe is already deployed, and in your opinion, might not be
the most architecturally sound solution, but we would need to factor
into account current deployments.=20

Best regards,
Milan

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of WORLEY, Dale R (Dale)
Sent: Friday, June 04, 2010 2:25 AM
To: Richard Shockey; 'Paul Kyzivat'; Patel, Milan
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Charter proposal for drafts related to the tel
URI

________________________________________
From: Richard Shockey [richard@shockey.us]

Though the larger issue of how to carry these types of messages or
metadata
(CNAM,CIC,SPID) in SIP has not been fully addressed either.

Needless to say we have had a vigorous discussion over this ongoing
issue in
both DRINKS and the E2MD BOF ( NGN ENUM ) list.
________________________________________

I suspect that any truly useful solution will be similar to what is
being solidified regarding translation Q.850 "response codes" into SIP
so that SIP can be used as a transit network between two SS7 networks:
(1) As much as possible, the *meaning* of the Q.850 code is translated
into a SIP response code, and (2) the original Q.850 code is carried
directly in a way for which SIP is transparent, in order to be
resurrected at a SIP-to-SS7 gateway.

In the case of CIC and friends, carrying the codes directly in a
suitable header is quite appropriate.  But I think that the approach of
attaching them directly to the originating or destination URI is not
architecturally sound.

Dale
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: "Patel, Milan" <Milan.Patel@InterDigital.com>, Richard Shockey <richard@shockey.us>, Paul Kyzivat <pkyzivat@cisco.com>, Cullen Jennings <fluffy@cisco.com>
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________________________________________
From: Patel, Milan [Milan.Patel@InterDigital.com]

Dale, the solutions currently proposed for the CPC/OLI/DAI is something
that I believe is already deployed, and in your opinion, might not be
the most architecturally sound solution, but we would need to factor
into account current deployments.
_______________________________________________

Beware that those who come to the IETF to get blessed things that they have=
 already deployed are often disappointed.  You cannot compel the IETF to ap=
prove things.

Dale

From pkyzivat@cisco.com  Fri Jun  4 06:37:08 2010
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Subject: Re: [dispatch] Charter proposal for drafts related to the tel URI
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Patel, Milan wrote:

> Dale, the solutions currently proposed for the CPC/OLI/DAI is something
> that I believe is already deployed, and in your opinion, might not be
> the most architecturally sound solution, but we would need to factor
> into account current deployments. 

We can certainly take that into account. But we don't rubber stamp 
everything that comes along that has been deployed. Architectural 
soundness is a very important consideration.

	Thanks,
	Paul

> Best regards,
> Milan
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of WORLEY, Dale R (Dale)
> Sent: Friday, June 04, 2010 2:25 AM
> To: Richard Shockey; 'Paul Kyzivat'; Patel, Milan
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Charter proposal for drafts related to the tel
> URI
> 
> ________________________________________
> From: Richard Shockey [richard@shockey.us]
> 
> Though the larger issue of how to carry these types of messages or
> metadata
> (CNAM,CIC,SPID) in SIP has not been fully addressed either.
> 
> Needless to say we have had a vigorous discussion over this ongoing
> issue in
> both DRINKS and the E2MD BOF ( NGN ENUM ) list.
> ________________________________________
> 
> I suspect that any truly useful solution will be similar to what is
> being solidified regarding translation Q.850 "response codes" into SIP
> so that SIP can be used as a transit network between two SS7 networks:
> (1) As much as possible, the *meaning* of the Q.850 code is translated
> into a SIP response code, and (2) the original Q.850 code is carried
> directly in a way for which SIP is transparent, in order to be
> resurrected at a SIP-to-SS7 gateway.
> 
> In the case of CIC and friends, carrying the codes directly in a
> suitable header is quite appropriate.  But I think that the approach of
> attaching them directly to the originating or destination URI is not
> architecturally sound.
> 
> Dale
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From HKaplan@acmepacket.com  Mon Jun  7 08:19:45 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 7 Jun 2010 09:51:03 -0400
Thread-Topic: Final Charter for Session-ID?
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Howdy,
I have heard no objections to this version of the charter (other than a deb=
ate about how to solve it). =20
If you have any comments regarding the charter, please speak now.  Thanks!

-hadriel


SIP Signaling-COmmunication Troubleshooting with Correlation Header (SIPSCO=
TCH) Charter:
----------------------------------
The SIPSCOTCH working group is chartered to define a mechanism that allows =
operators and monitoring equipment to correlate SIP messages and dialogs of=
 the same higher-level end-to-end "session" across multiple SIP device hops=
 for the same "message-flow", for the purpose of troubleshooting.  In parti=
cular, the mechanism must be able to allow such correlation across SIP B2BU=
As of as many functional types as possible, such as Application Servers, So=
ftswitches, PBXs, SBCs, etc.

The definition of a higher-layer "session" of the same "message-flow" is no=
t defined by SIP when it involves a B2BUA, and is difficult to normatively =
define in practice.  For the purposes of this WG charter's scope, two or mo=
re dialogs of a B2BUA are correlated under any conditions where correlation=
 might aid troubleshooting, by identifying them as belonging to the same hi=
gher-level session.  Any solution documents from this working group may exp=
and or constrain the scope of their mechanism's correlation, if the working=
 group so decides.

Although other solutions may be possible, this working group will focus on =
defining a new SIP Header field for such a purpose, similar to the Call-ID =
header field, in order to provide a simple, easily deployable mechanism whi=
ch can work across SIP domains. The SIP Call-ID header field value itself i=
s not usable for such correlation, because many B2BUAs must create a new Ca=
ll-ID value on their UAC "side" in order to function properly, and to compl=
y with the rules of RFC 3261.=20

It should be noted that certain SIP B2BUAs perform a "topology-hiding" func=
tion, as described in RFC 5853.  The working group will take such functions=
 into account for the correlation mechanism, as well as provide guidance fo=
r implementers of the mechanism to avoid triggering such a function for the=
 mechanism's header field value.

Lastly, although in certain environments such a mechanism may be usable for=
 dialog correlation in SIP state machines, it is not the goal of this worki=
ng group to address that usage.

Goals and Milestones
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Dec 2010 - New header specification to IESG (PS)



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Hash: SHA1

On 06/02/2010 02:02 PM, Peter Musgrave wrote:
>> Hi,
>>
>>  I support this proposed work to be taken by IETF under a new working group.  There are certainly concrete deliverables that would merit IETF consensus to spur multi-vendor >interoperability.
> 
> 
> +1

I also support this work.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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Testing list.

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Hi John,
Thank you for your comments:
Here are my view.

REQ-1:
I think that REQ-1 is still a justification, because not each Operator =
will use SIP-T for tunnelling purposes.
If you pass an untrusted domain SIP-T should not part, because it could =
include restricted numbers or other information that should not be =
presented to untrusted networks.

Also only for passing the Q.850 Cause code to encapsulate the whole ISUP =
is a little over dimensioned.
The SIP-T MIME can be lost due to the interdomain trust relation ships =
which the Reason header cannot.

For in an multi carrier/service provider environment like Germany is, it =
will not be secure that each operator will support SIP-T.=20

REQ-2
It must be possible to provide an accurate indication to a=20
UAC, so that the UAC can provide a suitable indication to the=20
user.

So that is the proposal for REQ-2.

REQ-3=20
   A UA may have the ability to display ISUP specific release causes or
   show a equivalent text.

So that would be sufficient enough for REQ-3. If people would like to =
see such text.

REQ-3/4
Your comment with regard to the type of UA is correct. I tried to =
satisfy people. The fact is that is not intended to include the Q.850 =
cause by an SIP end device. So the conclusion is changing REQ-3 and =
deleting REQ-4.

So now I would like to ask the people which requirements they would like =
to see in the draft.
As I understand John he sees only one (REQ-2)

I propose REQ-1, -2 and -3.

Opinions?

Thank you and Best Regards

Roland


Deutsche Telekom Netzproduktion GmbH
Zentrum Technik Einf=FChrung
Roland Jesske
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
+49 6151 628-2766 (Tel.)
+49 521 9210-3753  (Fax)
+49 171 8618-445  (Mobil)
E-Mail: mailto:r.jesske@telekom.de
http://www.telekom.com =20

Erleben, was verbindet.=20

Deutsche Telekom Netzproduktion GmbH
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr. DE 814645262

Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und nicht =
jede E-Mail drucken.

-----Urspr=FCngliche Nachricht-----
Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
Gesendet: Dienstag, 18. Mai 2010 21:37
An: Jesske, Roland; dispatch@ietf.org
Betreff: RE: Reason In =
responses(draft-jesske-dispatch-reason-in-responses-02)

Roland,=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: 18 May 2010 15:57
> To: Elwell, John; dispatch@ietf.org
> Subject: AW: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Hi John,
> Thank you for your comments.=20
> It was asked to split the draft into requirements and at=20
> least procedures.
>=20
> You are right that within the requirements draft is more that=20
> requirements. It is also the motivation, justification and=20
> examples. I can change the title if people are happy with it.=20
> Question is if such a draft is also needed as RFC or if it=20
> only seen as a temporary document during the discussion.
[JRE] I was not proposing a title change, and I have no problem =
combining motivation with requirements, although I do feel the examples =
look too much like solution and are unnecessary. Although some people =
asked you to split the draft, in my opinion it was wasted effort, but it =
is done now.

>=20
> With regard to REQ-1 I propose the following re-wording:
>=20
> It should be possible to support within native SIP=20
> environment PSTN-SIP-PSTN scenarios where the reason of a=20
> call release can be transferred though the SIP domain
> without any loss of information and no change of reason.
>=20
> I think this was discussed lengthy during the last meeting=20
> and email discussion.
> Not each PSTN GW will support ISUP MIME encapsulation. At=20
> least the 3GPP IMS has no SIP. The IMS is not using SIP-T.
[JRE] I don't think the new wording makes any difference. SIP-T is a =
means for tunnelling a Q.850 cause through a native SIP domain. The IETF =
already defines SIP-T as the solution, and if this solution is not =
sufficient, you need to cite reasons why not and identify a requirement =
that cannot be met using SIP-T. Just saying that IMS does not use SIP-T =
doesn't sound like a sufficient reason to me, because clearly IMS could =
be made to use SIP-T.

To be clear, I am not saying that the Reason header field should not be =
used in this situation, if the header field is defined for use in =
responses in general in order meet other requirements. I am simply =
saying that REQ-1 is not sufficient justification for allowing the =
Reason header field in responses, since REQ-1 can be satisfied using an =
existing mechanism.


>=20
> REQ-2 and REQ-3
> Yes it looks similar.
>=20
> REQ-3 was added because it was seen from user device=20
> perspective that the Device have the possibility to show the=20
> Q.850 cause but that such a End Device does not include Q.850=20
> causes. Of course a MGC should have this possibility.
>=20
> I would be happy to shrink REQ-2 and 3 to your proposed text.
>=20
> REQ-2
> It must be possible to provide an accurate indication to a=20
> UAC, so that the UAC can provide a suitable indication to the=20
> user. Whether this be an announcement, a textual message, an=20
> icon, a flashing lamp, a tone, a vibration, an odour or=20
> whatever, is a user interface matter.
[JRE] I did not intend the second sentence to be part of the requirement =
- it was just for illustration of why the existing text, which focused =
on textual display, was inadequate.

>=20
> REQ-4 was the consequence of REQ-3 where we do not allow the=20
> inclusion of a Q850 cause by a end device.
>=20
> So the trick is not allow the SIP end device to include a=20
> Q850 cause and in some certain cases to allow an service=20
> provider controlled application server to include a Reason=20
> header with a Q.850 cause. But it should not be done=20
> generally by SIP applications. So normally a ISUP=20
> implementation will use ISUP causes but not SIP implementations.
> But there are SIP Application servers out in the world that=20
> will have a ISUP application (or ISUP like application that=20
> reacts and behaves as within a ISUP network).
[JRE] I found REQ-3 rather confusing. On re-reading it, the first =
sentence is a requirement concerning reception, and the second sentence =
seems (if I understand correctly) to be a statement about sending. =
Assuming my understanding is correct, making a statement, within a =
requirement, that something is out of scope seems to suggest it is not =
part of the requirement.

In addition, the way SIP is specified, there is no real distinction =
between different types of UA. In other words, SIP procedures for UAs =
apply to UAs in general, not to specific types of UA like ASs. So =
writing a requirement that tries to distinguish between different types =
of UA does not seem to make sense.

>=20
> Question is now how to proceed further.
>=20
> Are people happy to have the documents split or put again=20
> together with striking out the examples ect?
> Or to have the documents as they are with some transfer of=20
> text to the draft-jesske-dispatch-reason-in-responses-02 draft?
[JRE] I don't have a strong opinion about document structure, although =
my personal preference would have been to leave it as one document, as =
it was originally. My main concern is trying to come up with a =
meaningful set of requirements, and in my opinion it reduces to a single =
requirement - being able to convey a Q.850 cause from PSTN (or similar) =
to a UAC so that the UAC can render meaningful information to the user.

John


>=20
> Comments?
>=20
> Thank you and Best Regards
>=20
> Roland
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Gesendet: Donnerstag, 13. Mai 2010 09:24
> An: Jesske, Roland; dispatch@ietf.org
> Betreff: RE: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Roland,
>=20
> In my view, there is a simple requirement to convey Q.850=20
> causes in SIP response messages, so that the UAC will receive=20
> a more accurate indication of the reason for rejection when=20
> rejection comes from PSTN. The cause mapping table helps to=20
> illustrate this. This is not rocket science, and I am=20
> disappointed that we have not found a quick way of=20
> dispatching this and just getting the work done. In my=20
> opinion the split into separate documents was unnecessary=20
> (take as an example the recently-published RFC 5876, which=20
> combines motivation and solution in one document), but it has=20
> been done now. Unfortunately I don't think it has moved us=20
> any further forward.
>=20
> I don't think the requirements in section 3 are particularly=20
> well expressed. REQ-1 can already be met by tunnelling in=20
> accordance with SIP-T. I would not expect the IETF to provide=20
> a new solution specifically for that requirement.
>=20
> I think REQ-2 and REQ-3 boil down to the same thing - it must=20
> be possible to provide an accurate indication to a UAC, so=20
> that the UAC can provide a suitable indication to the user.=20
> Whether this be an announcement, a textual message, an icon,=20
> a flashing lamp, a tone, a vibration, an odour or whatever,=20
> is a user interface matter.
>=20
> I also find REQ-4 rather problematic. If we have a solution=20
> for PSTN, we would also have a solution for some other form=20
> of gateway into a domain that provides a "PSTN-like service".=20
> Unfortunately, if you try to bring this out as a separate=20
> requirement, it raises the question of what exactly is this=20
> "PSTN-like service", and if it is somehow SIP-based, why it=20
> needs to be "PSTN-like", and so on.
>=20
> Basically I think there is only a single requirement here -=20
> provision of a correct indication to a UAC when rejection=20
> comes from the PSTN. I think that is a reasonable requirement=20
> and we should just go ahead and do it, rather like we did=20
> with RFC 5876 when we extended the range of messages that=20
> PAI/PPI could be used in.
>=20
> I find the statement in REQ-3 "A inclusion of [Q.850] causes=20
> is out of scope." very strange. I thought the whole idea was=20
> to convey Q.850 causes, so why say they are out of scope? I=20
> assume this is some sort of typo.
>=20
> I think one of the consequences of deciding to split=20
> requirements and solution into separate drafts is that the=20
> requirements draft has to stick to motivation and=20
> requirements. I find too much in the draft (particularly the=20
> examples) that smells of solution.
>=20
> John
>=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of R.Jesske@telekom.de
> > Sent: 13 May 2010 03:05
> > To: dispatch@ietf.org
> > Subject: Re: [dispatch] Reason In responses=20
> > (draft-jesske-dispatch-reason-in-responses-02)
> >=20
> > Dear all,=20
> > I have asked for comments on the two drafts mentioned below.=20
> > So far I didn't get any. Does that mean that people are happy=20
> > now with the content and we could proceed with it?=20
> >=20
> > Thank you and Best Regrads=20
> >=20
> > Roland=20
> >=20
> >=20
> > _____________________________________________=20
> > Von:    Jesske, Roland =20
> > Gesendet:       Donnerstag, 8. April 2010 09:46=20
> > An:     dispatch@ietf.org=20
> > Betreff:        Reason In responses=20
> > (draft-jesske-dispatch-reason-in-responses-02)=20
> >=20
> > Dear all,=20
> > During the discussion concerning the Reason header field in=20
> > Responses I was asked to split the document into an=20
> > requirements part and an protocol part.
> >=20
> > So please feel free to comment on the drafts.=20
> >=20
> > http://64.170.98.42/html/draft-jesske-dispatch-reason-in-respo
> > nses-02=20
> > <http://64.170.98.42/html/draft-jesske-dispatch-reason-in-resp
> > onses-02> =20
> >=20
> > http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-r
> > esponses-00=20
> > <http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-
> > responses-00> =20
> >=20
> > Best regards=20
> >=20
> > Roland Jesske=20
> >=20
> >=20
> > Deutsche Telekom Netzproduktion GmbH=20
> > Zentrum Technik Einf=FChrung=20
> > Roland Jesske=20
> > Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
> > +49 6151 628-2766 (Tel.)=20
> > +49 521 9210-3753  (Fax)=20
> > +49 171 8618-445  (Mobil)=20
> > E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de> =20
> > www.telekom.com <http:www.telekom.com>  =20
> >=20
> > Erleben, was verbindet.=20
> >=20
> > Deutsche Telekom Netzproduktion GmbH=20
> > Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)=20
> > Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> > Albert Matheis, Klaus Peren=20
> > Handelsregister: Amtsgericht Bonn HRB 14190=20
> > Sitz der Gesellschaft: Bonn=20
> > USt-IdNr. DE 814645262=20
> >=20
> > Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> > nicht jede E-Mail drucken.=20
> >=20
> >=20
> >=20
>=20

From Even.roni@huawei.com  Tue Jun  8 13:46:40 2010
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Date: Tue, 08 Jun 2010 23:43:45 +0300
From: Roni Even <Even.roni@huawei.com>
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To: "'Allyn Romanow (allyn)'" <allyn@cisco.com>, dispatch@ietf.org, mary.ietf.barnes@gmail.com, "'Cullen Jennings (fluffy)'" <fluffy@cisco.com>
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Subject: Re: [dispatch] Updated charter/work description for telepresence	multi-streams
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Hi,

I think this work is important and would like to see it addressed.

 

The first issue I see is that currently when the SDP offer includes multiple
m-lines of the same media type which typically will have different
semantics, it is not clear how the answerer will select the right streams
and render them correctly.  The content attribute provide some information
about the type of the stream but it does not discuss the application usage
in a way that will help interoperability. I believe that the use case and
requirements will provide more information about the interoperability issues
for this application and I think that this work will allow us to develop
similar solutions for other use cases that use multiple media stream of the
same media type .

 

Roni Even

 

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Allyn Romanow (allyn)
Sent: Wednesday, June 02, 2010 7:12 PM
To: dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen Jennings (fluffy)
Subject: [dispatch] Updated charter/work description for telepresence
multi-streams

 

Folks,

Here is a version of the work description that addresses the comments made
on the first draft. 

The changes include: 

1.       Charter section, qualification that we are working on SIP-based
systems

2.       Charter section, minor word smithing

3.       Charter section, reference to RAI WG with relevant work

4.       Scope section, clarification of interoperability requirements

5.       Scope section, replacement of second paragraph regarding treatment
of non-video non-audio media types

6.       Goals and Milestones - addition of a Problem Statement draft

 

 

Comments welcome

 

Allyn

 

Multi-streams for Telepresence Description of Work

 

 

 

Background

One branch of video conferencing has evolved that is focused on immersive
"being there" experience.  Referred to in various ways such as virtual
conferencing, telepresence or media spaces, early systems were mainly
research projects or business systems with limited deployments.  In recent
years telepresence systems have seen considerable market success.  Following
the model of early systems, the first wave of commercial systems have been
typically located in specially designed single-purpose rooms with multiple
relatively large displays permitting life size image reproduction, multiple
cameras, encoders, decoders and microphones.  These systems have several
important characteristics that are different from more traditional video
conferencing systems.  

 

The first difference concerns controlling the visual viewpoint in order to
improve participant nonverbal communication. These systems preserve
essential group meeting characteristics such as eye contact, group gestures,
seating order and spatial audio by carefully orchestrating the miking and
camera angles at each of the sites . This is distinct from the more
traditional approach where the geometric relationship between media streams
is not used to preserve inter-stream communication aspects such as eye
contact and group dynamics.  

 

A second difference is manipulation of the environment to improve immersion.
With telepresence systems, cinematographic aspects of the local environment
reproduction are carefully planned including color, table shape, seating and
lighting so that when combined with large high quality displays, a strong
sense of a "trompe l'oeil" or "being there" immersive experience is created.
Typical video conference systems do not include these considerations.

 

As telepresence video systems have become successful in the market,
manufacturers have started exploring delivery of the nonverbal communication
and immersive values of telepresence via smaller, less expensive and more
flexible video conferencing systems for a variety of venues, such as
individual offices, homes and kiosks. These are also  telepresence systems,
since the audio and video quality is high enough to allow clear image
reproduction for nonverbal communication, they are able to send and receive
multiple media streams, and large coordinated multi image displays are
available for immersive installations.   As the industry develops, the line
between telepresence and video conferencing may become blurred as nonverbal
communication and immersive installations become broadly available.

 

Problem

Although telepresence systems are based on open standards such as RTP, SIP,
H.264, H.323 suite, they cannot easily interoperate with each other without
operator assistance and expensive additional equipment that translates from
one vendor to another. It would be like having to make sure all parties are
on the same equipment (and network) when making a telephone call.  A major
factor in the inability of Telepresence systems to work with each other is
that there is no standard description of the multiple streams that comprise
the media flows. 

 

For example, in a multiple screen conference, the video and audio streams
sent from remote participants must be understood by receivers so that they
can be presented in a coherent and life-like manner. This includes the
ability to present the remote participants at their true size for their
apparent distance, while maintaining correct eye contact, gesticular cues,
and simultaneously providing a spatial audio sound stage consistent with the
video presentation.  The receiving device that decides how to display the
incoming information needs to understand a number of variables such as the
spatial position of the speaker, the field of view of the cameras, the
camera zoom, which media stream is related to each of the displays, etc. 

 

Charter

The Telepresence Multi-Streams work item in DISPATCH is chartered to define
and specify for SIP-based systems the content of media multi-stream messages
and the way these will be transported. 

 

This work will provide a standard for the exchange of media semantic
information that will foster interoperable end stations and conference
bridges. It will specify  variables that describe the semantics of the media
streams and the recommended behavior to achieve interoperability.  

 

This requires considering current widely deployed use cases, such as single
and multiple screens, multi-point, Scalable Video Coding (SVC), as well as
cases that are expected to be implemented using the protocol framework
produced by this work item.  The methodology for describing the variables
must allow extensibility of the variables, since telepresence is still a
young technology and may have use cases that are not currently considered.

 

The work item will identify use cases, define requirements, and define a
method for describing and transporting information about multiple media
streams, including a specification of variables required to support the use
cases. This work item will consider the reuse of existing IETF protocols and
produce an architecture/protocol framework document describing the protocols
required to be implemented to support this functionality.  The document will
identify any enhancements required to existing protocols as well as
describing new protocol(s) for interoperable multi-streams negotiation that
may be required.

 

Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and
FECframe.

 

Scope

The scope includes both systems that provide a fully immersive experience,
and systems that interwork with them and therefore need to understand the
same multiple stream semantics.

 

 

The focus of this work is on audio and video multiple streams.  Other media
types may be considered, however development of methodologies for them is
not within the scope of this work.

 

Interoperation with standards compliant systems is required, such as
SIP-based video conferencing systems.  However, backwards compatibility with
existing non-standards compliant telepresence systems is not required.

 

 

 

The group will produce

- Requirements and use cases

 

- Architectural Framework describing the protocols required to be
implemented to support this functionality and identifying existing protocol
enhancements and new protocol functionality required

 

- Specification of a new protocol to support telepresence multi-streams [if
required]

 

Goals and Milestones


Nov 2010 

 

Use Cases and Requirements to IESG as Informational RFC 


Nov 2010

March 2011 

Problem Statement to IESG as Informational RFC

Architecture to IESG as Informational RFC 


March 2011

Revise Charter [IF new protocol is not required]


Nov 2011 

Submit protocol draft to IESG as Proposed Standard RFC 

 

 

 


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>I think this work is =
important and
would like to see it addressed.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>The first issue I see =
is that
currently when the SDP offer includes multiple m-lines of the same media =
type which
typically will have different semantics, it is not clear how the =
answerer will
select the right streams and render them correctly. &nbsp;The content =
attribute
provide some information about the type of the stream but it does not =
discuss
the application usage in a way that will help interoperability. I =
believe that
the use case and requirements will provide more information about the =
interoperability
issues for this application and I think that this work will allow us to =
develop
similar solutions for other use cases that use multiple media stream of =
the
same media type .<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Roni =
Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>Allyn
Romanow (allyn)<br>
<b>Sent:</b> Wednesday, June 02, 2010 7:12 PM<br>
<b>To:</b> dispatch@ietf.org; mary.ietf.barnes@gmail.com; Cullen =
Jennings
(fluffy)<br>
<b>Subject:</b> [dispatch] Updated charter/work description for =
telepresence
multi-streams<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Folks,<o:p></o:p></p>

<p class=3DMsoNormal>Here is a version of the work description that =
addresses the
comments made on the first draft. <o:p></o:p></p>

<p class=3DMsoNormal>The changes include: <o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Charter section, =
qualification
that we are working on SIP-based systems<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Charter section, minor =
word
smithing<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Charter section, =
reference to RAI
WG with relevant work<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Scope section, =
clarification of
interoperability requirements<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>5.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Scope section, =
replacement of
second paragraph regarding treatment of non-video non-audio media =
types<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>6.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Goals and Milestones =
&#8211;
addition of a Problem Statement draft<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Comments welcome<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Allyn<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'>Multi-streams for =
Telepresence
Description of Work<o:p></o:p></span></b></p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><b><span
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></=
p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Background<o:p></o:p></span></=
b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>One branch of
video conferencing has evolved that is focused on immersive &#8220;being
there&#8221; experience.&nbsp; Referred to in various ways such as =
virtual conferencing,
telepresence or media spaces, early systems were mainly research =
projects or
business systems with limited deployments.&nbsp; In recent years =
telepresence
systems have seen considerable market success.&nbsp; Following the model =
of
early systems, the first wave of commercial systems have been typically =
located
in specially designed single-purpose rooms with multiple relatively =
large
displays permitting life size image reproduction, multiple cameras, =
encoders,
decoders and microphones.&nbsp; These systems have several important
characteristics that are different from more traditional video =
conferencing
systems.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The first
difference concerns controlling the visual viewpoint in order to improve
participant nonverbal communication. These systems preserve essential =
group
meeting characteristics such as eye contact, group gestures, seating =
order and
spatial audio by carefully orchestrating the miking and camera angles at =
each
of the sites . This is distinct from the more traditional approach where =
the
geometric relationship between media streams is not used to preserve
inter-stream communication aspects such as eye contact and group
dynamics.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>A =
second
difference is manipulation of the environment to improve =
immersion.&nbsp; With
telepresence systems, cinematographic aspects of the local environment
reproduction are carefully planned including color, table shape, seating =
and
lighting so that when combined with large high quality displays, a =
strong sense
of a &quot;trompe l'oeil&quot; or &quot;being there&quot; immersive =
experience
is created.&nbsp; Typical video conference systems do not include these
considerations.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>As
telepresence video systems have become successful in the market, =
manufacturers
have started exploring delivery of the nonverbal communication and =
immersive
values of telepresence via smaller, less expensive and more flexible =
video
conferencing systems for a variety of venues, such as individual =
offices, homes
and kiosks. These are also&nbsp; telepresence systems, since the audio =
and
video quality is high enough to allow clear image reproduction for =
nonverbal
communication, they are able to send and receive multiple media streams, =
and
large coordinated multi image displays are available for immersive
installations.&nbsp;&nbsp; As the industry develops, the line between
telepresence and video conferencing may become blurred as nonverbal
communication and immersive installations become broadly =
available.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Problem<o:p></o:p></span></b><=
/p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Although
telepresence systems are based on open standards such as RTP, SIP, =
H.264, H.323
suite, they cannot easily interoperate with each other without operator
assistance and expensive additional equipment that translates from one =
vendor
to another. It would be like having to make sure all parties are on the =
same
equipment (and network) when making a telephone call. &nbsp;A major =
factor in
the inability of Telepresence systems to work with each other is that =
there is
no standard description of the multiple streams that comprise the media =
flows. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>For example,
in a multiple screen conference, the video and audio streams sent from =
remote
participants must be understood by receivers so that they can be =
presented in a
coherent and life-like manner. This includes the ability to present the =
remote
participants at their true size for their apparent distance, while =
maintaining
correct eye contact, gesticular cues, and simultaneously providing a =
spatial
audio sound stage consistent with the video presentation.&nbsp; The =
receiving
device that decides how to display the incoming information needs to =
understand
a number of variables such as the spatial position of the speaker, the =
field of
view of the cameras, the camera zoom, which media stream is related to =
each of
the displays, etc. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Charter<o:p></o:p></span></b><=
/p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The
Telepresence Multi-Streams work item in DISPATCH is chartered to define =
and
specify for SIP-based systems the content of media multi-stream messages =
and
the way these will be transported. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>This
work will provide a standard for the exchange of media semantic =
information
that will foster interoperable end stations and conference bridges. It =
will
specify&nbsp; variables that describe the semantics of the media streams =
and
the recommended behavior to achieve interoperability.&nbsp; =
<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>This
requires considering current widely deployed use cases, such as single =
and
multiple screens, multi-point, Scalable Video Coding (SVC), as well as =
cases
that are expected to be implemented using the protocol framework =
produced by
this work item.&nbsp; The methodology for describing the variables must =
allow
extensibility of the variables, since telepresence is still a young =
technology
and may have use cases that are not currently =
considered.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>The
work item will identify use cases, define requirements, and define a =
method for
describing and transporting information about multiple media streams, =
including
a specification of variables required to support the use cases. This =
work item
will consider the reuse of existing IETF protocols and produce an
architecture/protocol framework document describing the protocols =
required to
be implemented to support this functionality.&nbsp; The document will =
identify
any enhancements required to existing protocols as well as describing =
new
protocol(s) for interoperable multi-streams negotiation that may be =
required.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>Relevant
work to be drawn upon has been done in XCON, MMUSIC, AVT, and =
FECframe.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><b><span =
style=3D'font-family:
"Arial","sans-serif"'>Scope<o:p></o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>The scope includes
both systems that provide a fully immersive experience, and systems that
interwork with them and therefore need to understand the same multiple =
stream
semantics.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'>The
focus of this work is on audio and video multiple streams.&nbsp; Other =
media
types may be considered, however development of methodologies for them =
is not
within the scope of this work.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:12.0pt;
font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>Interoperation
with standards compliant systems is required, such as SIP-based video
conferencing systems. &nbsp;However, backwards compatibility with =
existing
non-standards compliant telepresence systems is not =
required.<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><b><span =
style=3D'font-family:
"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:#0070C0'><o:p>&nbsp;</o:p=
></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>The group
will produce<o:p></o:p></span></b></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Requirements and use cases<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Architectural Framework describing the protocols required to be =
implemented to
support this functionality and identifying existing protocol =
enhancements and
new protocol functionality required<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-family:"Arial","sans-serif"'>-
Specification of a new protocol to support telepresence multi-streams =
[if
required]<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-family:"Arial","sans-serif"'>Goals and
Milestones<o:p></o:p></span></b></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2010 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Use Cases
  and Requirements to IESG as Informational RFC </span><span =
style=3D'font-size:
  12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2010</span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Problem
  Statement to IESG as Informational RFC</span><span =
style=3D'font-size:12.0pt;
  font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Architecture
  to IESG as Informational RFC </span><span =
style=3D'font-size:12.0pt;font-family:
  "Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>March 2011</span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Revise
  Charter [IF new protocol is not required]</span><span =
style=3D'font-size:12.0pt;
  font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
  </td>
 </tr>
 <tr>
  <td width=3D96 style=3D'width:1.0in;padding:.75pt .75pt .75pt .75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Nov 2011 </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
  <td width=3D381 style=3D'width:285.75pt;padding:.75pt .75pt .75pt =
.75pt'>
  <p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Submit
  protocol draft to IESG as Proposed Standard RFC </span><span
  =
style=3D'font-size:12.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</body>

</html>

--Boundary_(ID_76NPZnPMVDGGYmoMyNOytQ)--

From mary.ietf.barnes@gmail.com  Tue Jun  8 14:44:01 2010
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Cc: Robert Sparks <RjS@nostrum.com>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: [dispatch] DISPATCH-78 topics
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Hi all,

Per the reminder posted on the ML on May 4th and as listed on the
DISPATCH WG wiki, charter proposals for the topics to be
handled/dispatched prior to and at IETF-78 were due on May 31st.

We received charter proposals (problem statements/deliverables) for the
following, with links to most recent charter proposals and discussion threa=
ds.

o Session-ID: =A0updated charter proposal available
http://www.ietf.org/mail-archive/web/dispatch/current/msg01967.html

o Telepresence: =A0updated charter proposal available
http://www.ietf.org/mail-archive/web/dispatch/current/msg01936.html

o VIPR: =A0charter proposal
http://www.ietf.org/mail-archive/web/dispatch/current/msg01930.html

o TEL URI WG proposal: =A0discussion to focus on problem statement and scop=
e
http://www.ietf.org/mail-archive/web/dispatch/current/msg01920.html

The above items are the current targets for IETF-78 discussions. =A0Based
on those discussions, agenda time will be allocated, items dispatched
and adhocs scheduled as appropriate. Note, that the minimum time we
would allocate to a topic is 30 minutes and some may warrant 45
minutes. If we schedule adhocs, we will try to have those announced
around the time the agenda is finalized.

The following items have also been discussed on the mailing list, but
are not yet ready for dispatching:

o NGN Reason: Some interest. Needs more discussion and scoping.
http://datatracker.ietf.org/doc/draft-jesske-dispatch-reason-in-responses/
http://datatracker.ietf.org/doc/draft-jesske-dispatch-req-reason-in-respons=
es/

o PAN - wait for more energy
http://datatracker.ietf.org/doc/draft-lawrence-sipforum-provider-alias/

o =A0ACH - ACH analysis draft to happen as part of BLISS wind down
http://datatracker.ietf.org/doc/draft-ietf-bliss-ach-analysis/

o =A0ACH - HTTP API docs - need some hallway discussions
http://datatracker.ietf.org/doc/draft-yusef-dispatch-ach-rest-api/

o =A0Media security - needs additional ML discussion
http://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-parameter/

o Communication diversion and barring notification - feedback thus far
needs to be considered - i.e, problem statement to be clarified and
alternate solutions to be consider (Note: IPR on the communication
diversion draft)
http://datatracker.ietf.org/doc/draft-avasarala-dispatch-comm-div-notificat=
ion/
http://datatracker.ietf.org/doc/-avasarala-dispatch-comm-barring-notificati=
on/


As a reminder, the following are the cutoffs for drafts, so please make
sure that any drafts relevant to the topic are submitted prior to those
deadlines:
* - 00 documents: =A0July 5th (< 4 weeks)
* all other documents: July 12th (< 5 weeks)

Please keep in mind that the focus of discussions should be the problem
statement, scope and proposed deliverables for the topic.

As a reminder, items can be dispatched without being discussed at a f2f
meeting and the most effective way to achieve this is to have problem
statements, scope of topics and any relevant documents available for
discussion, rather than documents focusing on solutions.

Please, let the chairs know if there are any concerns.

Thanks,
Mary Barnes and Cullen Jennings
DISPATCH WG chairs

From alex@vidyo.com  Wed Jun  9 00:10:27 2010
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From: Alex Eleftheriadis <alex@vidyo.com>
To: "Allyn Romanow (allyn)" <allyn@cisco.com>
Date: Wed, 9 Jun 2010 03:10:15 -0400
Thread-Topic: [dispatch] Updated charter/work description fortelepresencemulti-streams
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Sure, that's fine.=20

--Alex

On Jun 3, 2010, at 3:53 AM, Allyn Romanow (allyn) wrote:

> Hi Alex and Hakon,=20
>=20
> I think the intent here is to say that we want to cover current use cases=
 and be extensible enough to cover new ones that may arise. I don't think i=
t valuable to exhaustively enumerate relevant dimensions of use cases in th=
is charter statement - that is better left to the use case document.=20
>=20
> How would it be to say,
>=20
> "This requires considering current widely deployed use cases, as well as =
cases that are expected to be implemented using the protocol framework prod=
uced by this work item.  The methodology for describing the variables must =
allow extensibility of the variables, since telepresence is still a young t=
echnology and may have use cases that are not currently considered."
>=20
>=20
> Allyn
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf Of H=E5kon Dahle
> Sent: Wednesday, June 02, 2010 5:02 PM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Updated charter/work description fortelepresencem=
ulti-streams
>=20
>=20
> Agree with Alex to a large extent in that  scalability, simulcast, publis=
h/subscribe may all be applicable to the problem at hand.
>=20
> However I do not think the Charter should jump to conclusions on what hap=
pens inside "the server" that Alex is referring to in his email - at this p=
oint I suggest leaving these details out of the charter, to make sure all o=
ptions can be investigated.
>=20
> -h
>=20
>=20
>=20
> ----- Reply message -----
> From: "Alex Eleftheriadis" <alex@vidyo.com>
> Date: Wed, Jun 2, 2010 16:20
> Subject: [dispatch] Updated charter/work description for telepresencemult=
i-streams
> To: "H=E5kon Dahle" <hakon.dahle@tandberg.com>
> Cc: "dispatch@ietf.org" <dispatch@ietf.org>
>=20
> Inline.
>=20
> On Jun 2, 2010, at 9:51 PM, H=E5kon Dahle wrote:
>=20
> A comment on the following sentence in the updated charter:
>=20
> =93This requires considering current widely deployed use cases, such as s=
ingle and multiple screens, multi-point, Scalable Video Coding (SVC), as we=
ll as cases that are expected to be implemented using the protocol framewor=
k produced by this work item.  =94
>=20
> Addressing use cases such as single and multiple screen systems, as well =
as both point-to-point and multi-point calling is great.  However I do not =
understand why a specific video codec or coding technique has been added to=
 the same context.  The charter shouldn=92t recommend or limit the work for=
 any specific codec (it should be codec agnostic),  and I suggest the chart=
er should be changed to comply with this.
>=20
> Right. I believe that the correct reference is to alternative architectur=
es for the server in both point-to-point and multi-point calls, where scala=
bility (which includes simulcasting, pyramidal coding such as SVC, and mult=
iple description coding) lends itself to VRUs (Video Routing Units) as oppo=
sed to MCUs. The VRU employs a "subscribe" model, made possible by scalabil=
ity, and which works quite differently than an MCU.
>=20
> --Alex
>=20
>=20
>=20
> -h
>=20
> ________________________________
> Fra: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> p=E5 veg=
ne av Allyn Romanow (allyn)
> Sendt: on 02.06.2010 18:11
> Til: dispatch@ietf.org<mailto:dispatch@ietf.org>; mary.ietf.barnes@gmail.=
com<mailto:mary.ietf.barnes@gmail.com>; Cullen Jennings (fluffy)
> Emne: [dispatch] Updated charter/work description for telepresencemulti-s=
treams
> Folks,
> Here is a version of the work description that addresses the comments mad=
e on the first draft.
> The changes include:
> 1.       Charter section, qualification that we are working on SIP-based =
systems
> 2.       Charter section, minor word smithing
> 3.       Charter section, reference to RAI WG with relevant work
> 4.       Scope section, clarification of interoperability requirements
> 5.       Scope section, replacement of second paragraph regarding treatme=
nt of non-video non-audio media types
> 6.       Goals and Milestones =96 addition of a Problem Statement draft
>=20
>=20
> Comments welcome
>=20
> Allyn
>=20
> Multi-streams for Telepresence Description of Work
>=20
>=20
>=20
> Background
> One branch of video conferencing has evolved that is focused on immersive=
 =93being there=94 experience.  Referred to in various ways such as virtual=
 conferencing, telepresence or media spaces, early systems were mainly rese=
arch projects or business systems with limited deployments.  In recent year=
s telepresence systems have seen considerable market success.  Following th=
e model of early systems, the first wave of commercial systems have been ty=
pically located in specially designed single-purpose rooms with multiple re=
latively large displays permitting life size image reproduction, multiple c=
ameras, encoders, decoders and microphones.  These systems have several imp=
ortant characteristics that are different from more traditional video confe=
rencing systems.
>=20
> The first difference concerns controlling the visual viewpoint in order t=
o improve participant nonverbal communication. These systems preserve essen=
tial group meeting characteristics such as eye contact, group gestures, sea=
ting order and spatial audio by carefully orchestrating the miking and came=
ra angles at each of the sites . This is distinct from the more traditional=
 approach where the geometric relationship between media streams is not use=
d to preserve inter-stream communication aspects such as eye contact and gr=
oup dynamics.
>=20
> A second difference is manipulation of the environment to improve immersi=
on.  With telepresence systems, cinematographic aspects of the local enviro=
nment reproduction are carefully planned including color, table shape, seat=
ing and lighting so that when combined with large high quality displays, a =
strong sense of a "trompe l'oeil" or "being there" immersive experience is =
created.  Typical video conference systems do not include these considerati=
ons.
>=20
> As telepresence video systems have become successful in the market, manuf=
acturers have started exploring delivery of the nonverbal communication and=
 immersive values of telepresence via smaller, less expensive and more flex=
ible video conferencing systems for a variety of venues, such as individual=
 offices, homes and kiosks. These are also  telepresence systems, since the=
 audio and video quality is high enough to allow clear image reproduction f=
or nonverbal communication, they are able to send and receive multiple medi=
a streams, and large coordinated multi image displays are available for imm=
ersive installations.   As the industry develops, the line between telepres=
ence and video conferencing may become blurred as nonverbal communication a=
nd immersive installations become broadly available.
>=20
> Problem
> Although telepresence systems are based on open standards such as RTP, SI=
P, H.264, H.323 suite, they cannot easily interoperate with each other with=
out operator assistance and expensive additional equipment that translates =
from one vendor to another. It would be like having to make sure all partie=
s are on the same equipment (and network) when making a telephone call.  A =
major factor in the inability of Telepresence systems to work with each oth=
er is that there is no standard description of the multiple streams that co=
mprise the media flows.
>=20
> For example, in a multiple screen conference, the video and audio streams=
 sent from remote participants must be understood by receivers so that they=
 can be presented in a coherent and life-like manner. This includes the abi=
lity to present the remote participants at their true size for their appare=
nt distance, while maintaining correct eye contact, gesticular cues, and si=
multaneously providing a spatial audio sound stage consistent with the vide=
o presentation.  The receiving device that decides how to display the incom=
ing information needs to understand a number of variables such as the spati=
al position of the speaker, the field of view of the cameras, the camera zo=
om, which media stream is related to each of the displays, etc.
>=20
> Charter
> The Telepresence Multi-Streams work item in DISPATCH is chartered to defi=
ne and specify for SIP-based systems the content of media multi-stream mess=
ages and the way these will be transported.
>=20
> This work will provide a standard for the exchange of media semantic info=
rmation that will foster interoperable end stations and conference bridges.=
 It will specify  variables that describe the semantics of the media stream=
s and the recommended behavior to achieve interoperability.
>=20
> This requires considering current widely deployed use cases, such as sing=
le and multiple screens, multi-point, Scalable Video Coding (SVC), as well =
as cases that are expected to be implemented using the protocol framework p=
roduced by this work item.  The methodology for describing the variables mu=
st allow extensibility of the variables, since telepresence is still a youn=
g technology and may have use cases that are not currently considered.
>=20
> The work item will identify use cases, define requirements, and define a =
method for describing and transporting information about multiple media str=
eams, including a specification of variables required to support the use ca=
ses. This work item will consider the reuse of existing IETF protocols and =
produce an architecture/protocol framework document describing the protocol=
s required to be implemented to support this functionality.  The document w=
ill identify any enhancements required to existing protocols as well as des=
cribing new protocol(s) for interoperable multi-streams negotiation that ma=
y be required.
>=20
> Relevant work to be drawn upon has been done in XCON, MMUSIC, AVT, and FE=
Cframe.
>=20
> Scope
> The scope includes both systems that provide a fully immersive experience=
, and systems that interwork with them and therefore need to understand the=
 same multiple stream semantics.
>=20
>=20
> The focus of this work is on audio and video multiple streams.  Other med=
ia types may be considered, however development of methodologies for them i=
s not within the scope of this work.
>=20
> Interoperation with standards compliant systems is required, such as SIP-=
based video conferencing systems.  However, backwards compatibility with ex=
isting non-standards compliant telepresence systems is not required.
>=20
>=20
>=20
> The group will produce
> - Requirements and use cases
>=20
> - Architectural Framework describing the protocols required to be impleme=
nted to support this functionality and identifying existing protocol enhanc=
ements and new protocol functionality required
>=20
> - Specification of a new protocol to support telepresence multi-streams [=
if required]
>=20
> Goals and Milestones
> Nov 2010
>=20
>=20
> Use Cases and Requirements to IESG as Informational RFC
>=20
> Nov 2010
> March 2011
>=20
> Problem Statement to IESG as Informational RFC
> Architecture to IESG as Informational RFC
>=20
> March 2011
>=20
> Revise Charter [IF new protocol is not required]
>=20
> Nov 2011
>=20
> Submit protocol draft to IESG as Proposed Standard RFC
>=20
>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org<mailto:dispatch@ietf.org>
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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From: Laura Liess <laura.liess.dt@googlemail.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>,  Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>,  "Worley, Dale R (Dale)" <dworley@avaya.com>
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Cc: Robert Sparks <RjS@nostrum.com>, DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] DISPATCH-78 topics
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Hi Mary, Gonzalo,

What about the Alert-Info URN?  We had a lot of discussion about the
charter and name, my opinion was that we had consensus on the charter.
Dale would chair the WG.  I intend to submit a new draft version by
the end of June.
I also sent to you an e-mail some time ago asking for a time slot for
discussion in Masstricht. Is it anything I missed to do to get agenda
time?

Thanks a lot
Laura




2010/6/8 Mary Barnes <mary.ietf.barnes@gmail.com>:
> Hi all,
>
> Per the reminder posted on the ML on May 4th and as listed on the
> DISPATCH WG wiki, charter proposals for the topics to be
> handled/dispatched prior to and at IETF-78 were due on May 31st.
>
> We received charter proposals (problem statements/deliverables) for the
> following, with links to most recent charter proposals and discussion thr=
eads.
>
> o Session-ID: =A0updated charter proposal available
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01967.html
>
> o Telepresence: =A0updated charter proposal available
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01936.html
>
> o VIPR: =A0charter proposal
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01930.html
>
> o TEL URI WG proposal: =A0discussion to focus on problem statement and sc=
ope
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01920.html
>
> The above items are the current targets for IETF-78 discussions. =A0Based
> on those discussions, agenda time will be allocated, items dispatched
> and adhocs scheduled as appropriate. Note, that the minimum time we
> would allocate to a topic is 30 minutes and some may warrant 45
> minutes. If we schedule adhocs, we will try to have those announced
> around the time the agenda is finalized.
>
> The following items have also been discussed on the mailing list, but
> are not yet ready for dispatching:
>
> o NGN Reason: Some interest. Needs more discussion and scoping.
> http://datatracker.ietf.org/doc/draft-jesske-dispatch-reason-in-responses=
/
> http://datatracker.ietf.org/doc/draft-jesske-dispatch-req-reason-in-respo=
nses/
>
> o PAN - wait for more energy
> http://datatracker.ietf.org/doc/draft-lawrence-sipforum-provider-alias/
>
> o =A0ACH - ACH analysis draft to happen as part of BLISS wind down
> http://datatracker.ietf.org/doc/draft-ietf-bliss-ach-analysis/
>
> o =A0ACH - HTTP API docs - need some hallway discussions
> http://datatracker.ietf.org/doc/draft-yusef-dispatch-ach-rest-api/
>
> o =A0Media security - needs additional ML discussion
> http://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-parameter/
>
> o Communication diversion and barring notification - feedback thus far
> needs to be considered - i.e, problem statement to be clarified and
> alternate solutions to be consider (Note: IPR on the communication
> diversion draft)
> http://datatracker.ietf.org/doc/draft-avasarala-dispatch-comm-div-notific=
ation/
> http://datatracker.ietf.org/doc/-avasarala-dispatch-comm-barring-notifica=
tion/
>
>
> As a reminder, the following are the cutoffs for drafts, so please make
> sure that any drafts relevant to the topic are submitted prior to those
> deadlines:
> * - 00 documents: =A0July 5th (< 4 weeks)
> * all other documents: July 12th (< 5 weeks)
>
> Please keep in mind that the focus of discussions should be the problem
> statement, scope and proposed deliverables for the topic.
>
> As a reminder, items can be dispatched without being discussed at a f2f
> meeting and the most effective way to achieve this is to have problem
> statements, scope of topics and any relevant documents available for
> discussion, rather than documents focusing on solutions.
>
> Please, let the chairs know if there are any concerns.
>
> Thanks,
> Mary Barnes and Cullen Jennings
> DISPATCH WG chairs
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From laura.liess.dt@googlemail.com  Wed Jun  9 00:33:06 2010
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+1
I think the charter is consistent to what is needed in the field.

Thanks
Laura


2010/6/7 Hadriel Kaplan <HKaplan@acmepacket.com>:
>
> Howdy,
> I have heard no objections to this version of the charter (other than a d=
ebate about how to solve it).
> If you have any comments regarding the charter, please speak now. =A0Than=
ks!
>
> -hadriel
>
>
> SIP Signaling-COmmunication Troubleshooting with Correlation Header (SIPS=
COTCH) Charter:
> ----------------------------------
> The SIPSCOTCH working group is chartered to define a mechanism that allow=
s operators and monitoring equipment to correlate SIP messages and dialogs =
of the same higher-level end-to-end "session" across multiple SIP device ho=
ps for the same "message-flow", for the purpose of troubleshooting. =A0In p=
articular, the mechanism must be able to allow such correlation across SIP =
B2BUAs of as many functional types as possible, such as Application Servers=
, Softswitches, PBXs, SBCs, etc.
>
> The definition of a higher-layer "session" of the same "message-flow" is =
not defined by SIP when it involves a B2BUA, and is difficult to normativel=
y define in practice. =A0For the purposes of this WG charter's scope, two o=
r more dialogs of a B2BUA are correlated under any conditions where correla=
tion might aid troubleshooting, by identifying them as belonging to the sam=
e higher-level session. =A0Any solution documents from this working group m=
ay expand or constrain the scope of their mechanism's correlation, if the w=
orking group so decides.
>
> Although other solutions may be possible, this working group will focus o=
n defining a new SIP Header field for such a purpose, similar to the Call-I=
D header field, in order to provide a simple, easily deployable mechanism w=
hich can work across SIP domains. The SIP Call-ID header field value itself=
 is not usable for such correlation, because many B2BUAs must create a new =
Call-ID value on their UAC "side" in order to function properly, and to com=
ply with the rules of RFC 3261.
>
> It should be noted that certain SIP B2BUAs perform a "topology-hiding" fu=
nction, as described in RFC 5853. =A0The working group will take such funct=
ions into account for the correlation mechanism, as well as provide guidanc=
e for implementers of the mechanism to avoid triggering such a function for=
 the mechanism's header field value.
>
> Lastly, although in certain environments such a mechanism may be usable f=
or dialog correlation in SIP state machines, it is not the goal of this wor=
king group to address that usage.
>
> Goals and Milestones
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Dec 2010 - New header specification to IESG (PS)
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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I am OK with both, too.

Thanks
Laura


2010/6/2 Cullen Jennings <fluffy@cisco.com>:
>
> +1 to ALES or SALUD
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From victor.pascual.avila@gmail.com  Wed Jun  9 02:26:13 2010
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On Mon, Jun 7, 2010 at 3:51 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wro=
te:
>
> Howdy,
> I have heard no objections to this version of the charter (other than a d=
ebate about how to solve it).
> If you have any comments regarding the charter, please speak now. =C2=A0T=
hanks!

I'm fine with this charter, it's addressing a real problem and the
scope is clear.

Cheers,
--=20
Victor Pascual =C3=81vila

From peter.musgrave@magorcorp.com  Wed Jun  9 02:47:16 2010
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I'd like to see this go forward too.

Peter

On Wed, Jun 9, 2010 at 5:26 AM, Victor Pascual Avila
<victor.pascual.avila@gmail.com> wrote:
> On Mon, Jun 7, 2010 at 3:51 PM, Hadriel Kaplan <HKaplan@acmepacket.com> w=
rote:
>>
>> Howdy,
>> I have heard no objections to this version of the charter (other than a =
debate about how to solve it).
>> If you have any comments regarding the charter, please speak now. =A0Tha=
nks!
>
> I'm fine with this charter, it's addressing a real problem and the
> scope is clear.
>
> Cheers,
> --
> Victor Pascual =C1vila
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From dyork@voxeo.com  Wed Jun  9 05:24:02 2010
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+1

--
Dan York
http://www.voxeo.com/
http://blogs.voxeo.com/
http://www.danyork.com/
http://twitter.com/danyork


On Jun 9, 2010, at 5:47 AM, Peter Musgrave =
<peter.musgrave@magorcorp.com> wrote:

> I'd like to see this go forward too.
>=20
> Peter
>=20
> On Wed, Jun 9, 2010 at 5:26 AM, Victor Pascual Avila
> <victor.pascual.avila@gmail.com> wrote:
>> On Mon, Jun 7, 2010 at 3:51 PM, Hadriel Kaplan =
<HKaplan@acmepacket.com> wrote:
>>>=20
>>> Howdy,
>>> I have heard no objections to this version of the charter (other =
than a debate about how to solve it).
>>> If you have any comments regarding the charter, please speak now.  =
Thanks!
>>=20
>> I'm fine with this charter, it's addressing a real problem and the
>> scope is clear.
>>=20
>> Cheers,
>> --
>> Victor Pascual =C3=81vila
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From dschwartz@xconnect.net  Wed Jun  9 05:26:36 2010
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From: David Schwartz <dschwartz@xconnect.net>
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Date: Wed, 9 Jun 2010 15:25:38 +0300
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+1

> On Wed, Jun 9, 2010 at 5:26 AM, Victor Pascual Avila
> <victor.pascual.avila@gmail.com> wrote:
>> On Mon, Jun 7, 2010 at 3:51 PM, Hadriel Kaplan <HKaplan@acmepacket.com> =
wrote:
>>>
>>> Howdy,
>>> I have heard no objections to this version of the charter (other than a=
 debate about how to solve it).
>>> If you have any comments regarding the charter, please speak now.  Than=
ks!
>>
>> I'm fine with this charter, it's addressing a real problem and the
>> scope is clear.
>>
>> Cheers,
>> --
>> Victor Pascual =C1vila
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From fluffy@cisco.com  Wed Jun  9 07:53:30 2010
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I have a problem with this proposed charter.=20

On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:

>=20
> The definition of a higher-layer "session" of the same "message-flow" =
is not defined by SIP when it involves a B2BUA, and is difficult to =
normatively define in practice.  For the purposes of this WG charter's =
scope, two or more dialogs of a B2BUA are correlated under any =
conditions where correlation might aid troubleshooting, by identifying =
them as belonging to the same higher-level session. Any solution =
documents from this working group may expand or constrain the scope of =
their mechanism's correlation, if the working group so decides.

I think the above text is way too vague and impossible to reduce to =
engineering practice. If you have 10 calls going to a PSTN GWs that is =
having voice quality problems, clearly having all of them grouped =
together is useful in debugging. Yet I doubt we want all the calls from =
a given PSTN GW to have the same correlation ID. Our lack of any solid =
definition of what it is we are trying to correlate is a strong =
indication to me that we have not yet come to any consensus about what =
the problem is this WG would be trying to fix.
=20
I would like to see a WG charted in this area but I would expect the =
IESG to more or less laugh at a problem statement that reduced down to =
"uh might be useful for debuggin". Can some people please try and =
propose a more concrete definition of when two sip dialogs will be =
correlated?

Cullen


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Cc: Robert Sparks <RjS@nostrum.com>, DISPATCH <dispatch@ietf.org>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [dispatch] DISPATCH-78 topics - Alert URNs
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Hi Laura,

We did not consider this topic for pre-IETF-78 focus for the DISPATCH
WG because the ADs are already in the process of preparing the charter
for approval by the IESG, as I understood Gonzalo's emails on the
topic.   I'll let ADs discuss the timeline for the chartering and
really it would be up to them as to whether a meeting agenda slot
would be requested for the topic.

The current model we've been using is that past topics that have been
agreed to be chartered have already been "dispatched" per se, per the
summaries in the wiki:
http://trac.tools.ietf.org/wg/dispatch/trac/wiki

Thanks,
Mary.




On Wed, Jun 9, 2010 at 2:19 AM, Laura Liess
<laura.liess.dt@googlemail.com> wrote:
> Hi Mary, Gonzalo,
>
> What about the Alert-Info URN? =A0We had a lot of discussion about the
> charter and name, my opinion was that we had consensus on the charter.
> Dale would chair the WG. =A0I intend to submit a new draft version by
> the end of June.
> I also sent to you an e-mail some time ago asking for a time slot for
> discussion in Masstricht. Is it anything I missed to do to get agenda
> time?
>
> Thanks a lot
> Laura
>
>
>
>
> 2010/6/8 Mary Barnes <mary.ietf.barnes@gmail.com>:
>> Hi all,
>>
>> Per the reminder posted on the ML on May 4th and as listed on the
>> DISPATCH WG wiki, charter proposals for the topics to be
>> handled/dispatched prior to and at IETF-78 were due on May 31st.
>>
>> We received charter proposals (problem statements/deliverables) for the
>> following, with links to most recent charter proposals and discussion th=
reads.
>>
>> o Session-ID: =A0updated charter proposal available
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01967.html
>>
>> o Telepresence: =A0updated charter proposal available
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01936.html
>>
>> o VIPR: =A0charter proposal
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01930.html
>>
>> o TEL URI WG proposal: =A0discussion to focus on problem statement and s=
cope
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01920.html
>>
>> The above items are the current targets for IETF-78 discussions. =A0Base=
d
>> on those discussions, agenda time will be allocated, items dispatched
>> and adhocs scheduled as appropriate. Note, that the minimum time we
>> would allocate to a topic is 30 minutes and some may warrant 45
>> minutes. If we schedule adhocs, we will try to have those announced
>> around the time the agenda is finalized.
>>
>> The following items have also been discussed on the mailing list, but
>> are not yet ready for dispatching:
>>
>> o NGN Reason: Some interest. Needs more discussion and scoping.
>> http://datatracker.ietf.org/doc/draft-jesske-dispatch-reason-in-response=
s/
>> http://datatracker.ietf.org/doc/draft-jesske-dispatch-req-reason-in-resp=
onses/
>>
>> o PAN - wait for more energy
>> http://datatracker.ietf.org/doc/draft-lawrence-sipforum-provider-alias/
>>
>> o =A0ACH - ACH analysis draft to happen as part of BLISS wind down
>> http://datatracker.ietf.org/doc/draft-ietf-bliss-ach-analysis/
>>
>> o =A0ACH - HTTP API docs - need some hallway discussions
>> http://datatracker.ietf.org/doc/draft-yusef-dispatch-ach-rest-api/
>>
>> o =A0Media security - needs additional ML discussion
>> http://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-parameter/
>>
>> o Communication diversion and barring notification - feedback thus far
>> needs to be considered - i.e, problem statement to be clarified and
>> alternate solutions to be consider (Note: IPR on the communication
>> diversion draft)
>> http://datatracker.ietf.org/doc/draft-avasarala-dispatch-comm-div-notifi=
cation/
>> http://datatracker.ietf.org/doc/-avasarala-dispatch-comm-barring-notific=
ation/
>>
>>
>> As a reminder, the following are the cutoffs for drafts, so please make
>> sure that any drafts relevant to the topic are submitted prior to those
>> deadlines:
>> * - 00 documents: =A0July 5th (< 4 weeks)
>> * all other documents: July 12th (< 5 weeks)
>>
>> Please keep in mind that the focus of discussions should be the problem
>> statement, scope and proposed deliverables for the topic.
>>
>> As a reminder, items can be dispatched without being discussed at a f2f
>> meeting and the most effective way to achieve this is to have problem
>> statements, scope of topics and any relevant documents available for
>> discussion, rather than documents focusing on solutions.
>>
>> Please, let the chairs know if there are any concerns.
>>
>> Thanks,
>> Mary Barnes and Cullen Jennings
>> DISPATCH WG chairs
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>

From mperumal@cisco.com  Wed Jun  9 23:07:55 2010
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From: "Muthu ArulMozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "DISPATCH list" <dispatch@ietf.org>
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Subject: Re: [dispatch] Final Charter for Session-ID?
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Here is an initial attempt:

If there is a dialog between A and B then A and B are in a session.
If there is a dialog between A and B and a dialog between B and C at the
same time then A, B and C are in a session.
More generally, if two or more participants are part of a session S and
at the same time there is a dialog between one of the participants of S
and another participant Z who is not part of S, then S (union) Z is a
session.

I think this would cover most of SBC, 3PCC an app server cases. People
can add cases like park/retrieve if they think they should be considered
as part of a single session.

Muthu

|-----Original Message-----
|From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
|Of Cullen Jennings (fluffy)
|Sent: Wednesday, June 09, 2010 8:23 PM
|To: DISPATCH list
|Subject: Re: [dispatch] Final Charter for Session-ID?
|
|I have a problem with this proposed charter.
|
|On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
|
|>
|> The definition of a higher-layer "session" of the same "message-flow"
is
|not defined by SIP when it involves a B2BUA, and is difficult to
normatively
|define in practice.  For the purposes of this WG charter's scope, two
or
|more dialogs of a B2BUA are correlated under any conditions where
|correlation might aid troubleshooting, by identifying them as belonging
to
|the same higher-level session. Any solution documents from this working
|group may expand or constrain the scope of their mechanism's
correlation, if
|the working group so decides.
|
|I think the above text is way too vague and impossible to reduce to
|engineering practice. If you have 10 calls going to a PSTN GWs that is
|having voice quality problems, clearly having all of them grouped
together
|is useful in debugging. Yet I doubt we want all the calls from a given
PSTN
|GW to have the same correlation ID. Our lack of any solid definition of
what
|it is we are trying to correlate is a strong indication to me that we
have
|not yet come to any consensus about what the problem is this WG would
be
|trying to fix.
|
|I would like to see a WG charted in this area but I would expect the
IESG to
|more or less laugh at a problem statement that reduced down to "uh
might be
|useful for debuggin". Can some people please try and propose a more
concrete
|definition of when two sip dialogs will be correlated?
|
|Cullen
|
|_______________________________________________
|dispatch mailing list
|dispatch@ietf.org
|https://www.ietf.org/mailman/listinfo/dispatch

From L.Liess@telekom.de  Wed Jun  9 23:11:56 2010
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Hi Mary,

Thanks a lot.

Laura
=20

>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>Sent: Wednesday, June 09, 2010 7:13 PM
>To: Laura Liess
>Cc: Robert Sparks; DISPATCH; Gonzalo Camarillo
>Subject: Re: [dispatch] DISPATCH-78 topics - Alert URNs
>
>Hi Laura,
>
>We did not consider this topic for pre-IETF-78 focus for the DISPATCH
>WG because the ADs are already in the process of preparing the charter
>for approval by the IESG, as I understood Gonzalo's emails on the
>topic.   I'll let ADs discuss the timeline for the chartering and
>really it would be up to them as to whether a meeting agenda slot
>would be requested for the topic.
>
>The current model we've been using is that past topics that have been
>agreed to be chartered have already been "dispatched" per se, per the
>summaries in the wiki:
>http://trac.tools.ietf.org/wg/dispatch/trac/wiki
>
>Thanks,
>Mary.
>
>
>
>
>On Wed, Jun 9, 2010 at 2:19 AM, Laura Liess
><laura.liess.dt@googlemail.com> wrote:
>> Hi Mary, Gonzalo,
>>
>> What about the Alert-Info URN? =A0We had a lot of discussion about =
the
>> charter and name, my opinion was that we had consensus on=20
>the charter.
>> Dale would chair the WG. =A0I intend to submit a new draft version by
>> the end of June.
>> I also sent to you an e-mail some time ago asking for a time slot for
>> discussion in Masstricht. Is it anything I missed to do to get agenda
>> time?
>>
>> Thanks a lot
>> Laura
>>
>>
>>
>>
>> 2010/6/8 Mary Barnes <mary.ietf.barnes@gmail.com>:
>>> Hi all,
>>>
>>> Per the reminder posted on the ML on May 4th and as listed on the
>>> DISPATCH WG wiki, charter proposals for the topics to be
>>> handled/dispatched prior to and at IETF-78 were due on May 31st.
>>>
>>> We received charter proposals (problem=20
>statements/deliverables) for the
>>> following, with links to most recent charter proposals and=20
>discussion threads.
>>>
>>> o Session-ID: =A0updated charter proposal available
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01967.html
>>>
>>> o Telepresence: =A0updated charter proposal available
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01936.html
>>>
>>> o VIPR: =A0charter proposal
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01930.html
>>>
>>> o TEL URI WG proposal: =A0discussion to focus on problem=20
>statement and scope
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01920.html
>>>
>>> The above items are the current targets for IETF-78=20
>discussions. =A0Based
>>> on those discussions, agenda time will be allocated, items=20
>dispatched
>>> and adhocs scheduled as appropriate. Note, that the minimum time we
>>> would allocate to a topic is 30 minutes and some may warrant 45
>>> minutes. If we schedule adhocs, we will try to have those announced
>>> around the time the agenda is finalized.
>>>
>>> The following items have also been discussed on the mailing=20
>list, but
>>> are not yet ready for dispatching:
>>>
>>> o NGN Reason: Some interest. Needs more discussion and scoping.
>>>=20
>http://datatracker.ietf.org/doc/draft-jesske-dispatch-reason-in
>-responses/
>>>=20
>http://datatracker.ietf.org/doc/draft-jesske-dispatch-req-reaso
n-in-responses/
>>>
>>> o PAN - wait for more energy
>>>=20
>http://datatracker.ietf.org/doc/draft-lawrence-sipforum-provider-alias/
>>>
>>> o =A0ACH - ACH analysis draft to happen as part of BLISS wind down
>>> http://datatracker.ietf.org/doc/draft-ietf-bliss-ach-analysis/
>>>
>>> o =A0ACH - HTTP API docs - need some hallway discussions
>>> http://datatracker.ietf.org/doc/draft-yusef-dispatch-ach-rest-api/
>>>
>>> o =A0Media security - needs additional ML discussion
>>>=20
>http://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-p
arameter/
>>>
>>> o Communication diversion and barring notification -=20
>feedback thus far
>>> needs to be considered - i.e, problem statement to be clarified and
>>> alternate solutions to be consider (Note: IPR on the communication
>>> diversion draft)
>>>=20
>http://datatracker.ietf.org/doc/draft-avasarala-dispatch-comm-d
iv-notification/
>>>=20
>http://datatracker.ietf.org/doc/-avasarala-dispatch-comm-barrin
g-notification/
>>>
>>>
>>> As a reminder, the following are the cutoffs for drafts, so=20
>please make
>>> sure that any drafts relevant to the topic are submitted=20
>prior to those
>>> deadlines:
>>> * - 00 documents: =A0July 5th (< 4 weeks)
>>> * all other documents: July 12th (< 5 weeks)
>>>
>>> Please keep in mind that the focus of discussions should be=20
>the problem
>>> statement, scope and proposed deliverables for the topic.
>>>
>>> As a reminder, items can be dispatched without being=20
>discussed at a f2f
>>> meeting and the most effective way to achieve this is to=20
>have problem
>>> statements, scope of topics and any relevant documents available for
>>> discussion, rather than documents focusing on solutions.
>>>
>>> Please, let the chairs know if there are any concerns.
>>>
>>> Thanks,
>>> Mary Barnes and Cullen Jennings
>>> DISPATCH WG chairs
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>

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Hello All,
I agree with Cullen in that the implementer of a B2BUA will need more =
guidance than is in the proposed charter now in order to decide whether =
dialogs are related, but I think the first question is how much of this =
guidance goes in the charter and how much in the working group drafts.

It might help to consider what it means for a UA to provide a =
correlation header field (e.g. session-ID) at dialog setup. Using the =
transfer case from Paul below:

>> 	A ----------- B ----------- C
>> 	              |
>> 	              |------------ D

If A adds a session-ID header field, then it means that A is interested =
in the successful functioning of this session, and B2BUA B has this =
knowledge when it receives an INVITE from A containing the session-ID =
header field. At transfer of C to D, the session-ID value shall be the =
same since A is still in the session. If the transfer is done by D =
sending an INVITE, then B2BUA B can insert the session-ID. I don't think =
it matters that D doesn't put the session-ID header field in its INVITE =
request since correlation is interested in what happens to A, not in the =
ability of D to enter a session or the ability of C to transfer a =
session.

Regarding the wording of the charter, I think only one of the two =
undefined terms "higher-layer session" and "message-flow" should be used =
(i.e. replace "higher-layer session" with "message-flow" or vice versa).

A possible revision of the charter text is below.

Regards,
Peter

SIP Signaling-COmmunication Troubleshooting with Correlation Header =
(SIPSCOTCH) Charter:
----------------------------------
The SIPSCOTCH working group is chartered to define a mechanism that =
allows operators and monitoring equipment to correlate multiple SIP =
messages and dialogs  across multiple SIP device hops for the purpose of =
troubleshooting in the case when such messages or dialogs provide
a single communication experience for a single user. A single user may =
be network entity such as a SIP B2BUA, PBX,
Softswitch, etc.=A0In particular, the mechanism must be able to allow =
such correlation across SIP B2BUAs of as many functional types as =
possible, such as  Application Servers, Softswitches, PBXs, SBCs, etc.

For the purposes of this WG charter's scope, two or more dialogs of a =
B2BUA are correlated under any conditions where correlation might aid =
troubleshooting,  by identifying them as providing a single =
communication experience to a single user. A correlation identifier will =
be duplicated in each dialog that impacts  the communication experience =
of the user agent that provided that correlation identifier. Any =
solution documents from this working group may expand or  constrain the =
scope of their mechanism's correlation, if the working group so decides.

Although other solutions may be possible, this working group will focus =
on defining a new SIP Header field for such a purpose, similar to the =
Call-ID header  field, in order to provide a simple, easily deployable =
mechanism which can work across SIP domains. The SIP Call-ID header =
field value itself is not usable  for such correlation, because many =
B2BUAs must create a new Call-ID value on their UAC "side" in order to =
function properly, and to comply with the rules of  RFC 3261.

It should be noted that certain SIP B2BUAs perform a "topology-hiding" =
function, as described in RFC 5853. =A0The working group will take such =
functions into  account for the correlation mechanism, as well as =
provide guidance for implementers of the mechanism to avoid triggering =
such a function for the mechanism's  header field value.

Lastly, although in certain environments such a mechanism may be usable =
for dialog correlation in SIP state machines, it is not the goal of this =
working  group to address that usage.


-------------------------------------------------------------------------=
-
I have a problem with this proposed charter.=20

On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:

>=20
> The definition of a higher-layer "session" of the same "message-flow" =
is not defined by SIP when it involves a B2BUA, and is difficult to =
normatively define in practice.  For the purposes of this WG charter's =
scope, two or more dialogs of a B2BUA are correlated under any =
conditions where correlation might aid troubleshooting, by identifying =
them as belonging to the same higher-level session. Any solution =
documents from this working group may expand or constrain the scope of =
their mechanism's correlation, if the working group so decides.

I think the above text is way too vague and impossible to reduce to =
engineering practice. If you have 10 calls going to a PSTN GWs that is =
having voice quality problems, clearly having all of them grouped =
together is useful in debugging. Yet I doubt we want all the calls from =
a given PSTN GW to have the same correlation ID. Our lack of any solid =
definition of what it is we are trying to correlate is a strong =
indication to me that we have not yet come to any consensus about what =
the problem is this WG would be trying to fix.
=20
I would like to see a WG charted in this area but I would expect the =
IESG to more or less laugh at a problem statement that reduced down to =
"uh might be useful for debuggin". Can some people please try and =
propose a more concrete definition of when two sip dialogs will be =
correlated?

Cullen




Message: 5
Date: Sat, 15 May 2010 00:00:37 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [dispatch] Proposed update to Session-ID charter, round-4
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Cc: DISPATCH <dispatch@ietf.org>
Message-ID: <4BEE1C65.1090507@cisco.com>
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Hadriel,

I don't disagree with anything you say below.
But it is also what is troubling to me.

It seems there is no one obvious definition for which dialogs should be =
related - there are good reasons for each of several possible =
definitions. Yet its hard to imagine supporting them all. And leaving it =
to the discretion of each server seems even worse. So I don't see how to =
decide on a definition.

	Thanks,
	Paul

Hadriel Kaplan wrote:
>=20
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Friday, May 14, 2010 7:56 PM
>>
>> Hadriel Kaplan wrote:
>>> For media-based sessions, such B2BUA's actually can and do represent
>> different media sources at different points in the call; for example=20
>> app servers which generate specific tones/music to the UAC while they =

>> hunt for the called party.
>>
>> I find very little difference between this scenario and one where the =

>> call is first answered by a secretary and then transferred to the=20
>> intended callee. (Does it matter that the secretary is automated?)
>=20
> In the case I was describing, the first dialog leg doesn't reach an =
established state until the party is found.  If it *did*, which happens =
in some find-me-follow-me app servers, then I would think they're really =
separate session-ids.  I'm not super-tide to that position, just seems =
logical to me for some reason.
>=20
> =20
>> I'm a bit confused about how things should work if the transfer case=20
>> does *not* share the same session-id:
>>
>> 	A ----------- B ----------- C
>> 	              |
>> 	              |------------ D
>>
>> Initially from A via B2BUA B to C, and gets Session-ID X applied to=20
>> both the A-B leg and the B-C leg. Then B replaces the B-C "leg" with=20
>> the B-D "leg". If the B-D leg doesn't get session-id X, and instead=20
>> gets session-ID Y, does A-B then also get a new session-id of Y??? It =

>> seems it must if the correlation is to work for that.
>> That then means that a single dialog, A-B has more than one =
session-id:
>> X and Y. Is that what is intended???
>=20
> In the current Session-ID draft, it makes no statement about that, =
iirc. =20
> I think that's one of the things the SIPSCOTCH WG must decide: whether =
A-B-D is in fact the same session-id as A-B-C.  If it *is*, then I'd =
argue a transferred call from A directly to D, replacing A-B-C, is also =
the same session-id.
>=20
> I can see it both ways.  Arguably, if A-B-C got to the established =
state, that was higher-layer Call-1.  If B-C gets replaced with B-D, =
then A-B-D is Call-2.  On the other hand, from a troubleshooting =
perspective it would be nice to know those dialogs are all related.
>=20
> Note that once you open the door for the session-id to go beyond the =
initial established call, it can get weird and complicated.
>=20
> For example, what happens if you call a public-announcement =
broadcaster server; you press some special codes, record an =
announcement, and press "broadcast"; do the sessions it creates for you =
use the same session-id as your call?
>=20
> Or take your example above, B-D could have replaced B-C by *D* sending =

> an INVITE with replaces to *B*.  So we'd have to get the Session-ID=20
> value to D so it can use it in its INVITE to B, like by C passing it=20
> in the Refer-To of a REFER to D.  OK, so now imagine if B parked the=20
> call and Referred D to C, and you ended up with 2 dialogs A-B, C-D as=20
> below.  But they'd be using the same session-id.  Is that legit?=20
> (maybe, I dunno)
>=20
>  	A ----------- B             C
>  	                            |
>  	                            D
>=20
> -hadriel
>=20
>=20


------------------------------

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End of dispatch Digest, Vol 14, Issue 37
****************************************

Peter Dawes=20

From peter.musgrave@magorcorp.com  Thu Jun 10 03:15:29 2010
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Hi,

I also agree that this is a genuine problem. I do think that a few
user cases would cover a large proportion of the common real world
scenarios (auto-attendant, transfer, forward to voice mail etc.) but
that to specify this exhaustively will likely prove difficult.

Is there any support for the idea that the WG should be chartered to
investigate this issue and provide a doc which indicates how B2BUAs
should handle the common cases? Either in the session ID draft or [my
preference] in a second doc (so that e.g. session-id can be used by
sipclf before the the discussion on detailed usage is complete).

Peter Musgrave

From Peter.Dawes@vodafone.com  Thu Jun 10 03:27:48 2010
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Hello Hadriel, All,
I believe that the session-ID header field is very useful and should be
described in an RFC. However, for troubleshooting across many network
entities where the message path is not known beforehand,
http://www.ietf.org/id/draft-dawes-sipping-debug-02.txt describes a
targetted approach that avoids having to look for session-ID in all
messages everywhere in the network. I believe that my draft is a
specialization of session-ID and could be written as an extension to it.


Does the proposed charter text allow my draft to be added to SIPSCOTCH
working group discussions? If not, I would like to propose adding it to
the charter. Thanks.

Regards,
Peter

-------------------------------------------------------------------

Howdy,
I have heard no objections to this version of the charter (other than a
debate about how to solve it). =20
If you have any comments regarding the charter, please speak now.
Thanks!

-hadriel


SIP Signaling-COmmunication Troubleshooting with Correlation Header
(SIPSCOTCH) Charter:
----------------------------------
The SIPSCOTCH working group is chartered to define a mechanism that
allows operators and monitoring equipment to correlate SIP messages and
dialogs of the same higher-level end-to-end "session" across multiple
SIP device hops for the same "message-flow", for the purpose of
troubleshooting.  In particular, the mechanism must be able to allow
such correlation across SIP B2BUAs of as many functional types as
possible, such as Application Servers, Softswitches, PBXs, SBCs, etc.

The definition of a higher-layer "session" of the same "message-flow" is
not defined by SIP when it involves a B2BUA, and is difficult to
normatively define in practice.  For the purposes of this WG charter's
scope, two or more dialogs of a B2BUA are correlated under any
conditions where correlation might aid troubleshooting, by identifying
them as belonging to the same higher-level session.  Any solution
documents from this working group may expand or constrain the scope of
their mechanism's correlation, if the working group so decides.

Although other solutions may be possible, this working group will focus
on defining a new SIP Header field for such a purpose, similar to the
Call-ID header field, in order to provide a simple, easily deployable
mechanism which can work across SIP domains. The SIP Call-ID header
field value itself is not usable for such correlation, because many
B2BUAs must create a new Call-ID value on their UAC "side" in order to
function properly, and to comply with the rules of RFC 3261.=20

It should be noted that certain SIP B2BUAs perform a "topology-hiding"
function, as described in RFC 5853.  The working group will take such
functions into account for the correlation mechanism, as well as provide
guidance for implementers of the mechanism to avoid triggering such a
function for the mechanism's header field value.

Lastly, although in certain environments such a mechanism may be usable
for dialog correlation in SIP state machines, it is not the goal of this
working group to address that usage.

Goals and Milestones
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Dec 2010 - New header specification to IESG (PS)



Peter Dawes=20
Group R&D
=20
Tel: +44 (0)7717 275009
Fax: +44 (0)1635 234873

=20
E-mail: peter.dawes@vodafone.com
=20
www.betavine.net  - Web
betavine.mobi  - Mobile Web
=20
Vodafone Group Services Limited=20
Registered Office: Vodafone House, The Connection, Newbury, Berkshire
RG14 2FN Registered in England No 3802001


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Muthu ArulMozhi Perumal (mperumal) wrote:
> Here is an initial attempt:
> 
> If there is a dialog between A and B then A and B are in a session.
> If there is a dialog between A and B and a dialog between B and C at the
> same time then A, B and C are in a session.
> More generally, if two or more participants are part of a session S and
> at the same time there is a dialog between one of the participants of S
> and another participant Z who is not part of S, then S (union) Z is a
> session.

The above just reiterates the problem that Cullen was raising.
Just because B has a dialog with A and a dialog with C does not mean 
that there is a session between A and C. A gateway provides a perfect 
example of that.

There needs to be some semantic link between the dialogs. The question 
is defining what sorts of semantic links should be used to share a 
session id.

	Thanks,
	Paul

> I think this would cover most of SBC, 3PCC an app server cases. People
> can add cases like park/retrieve if they think they should be considered
> as part of a single session.
> 
> Muthu
> 
> |-----Original Message-----
> |From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf
> |Of Cullen Jennings (fluffy)
> |Sent: Wednesday, June 09, 2010 8:23 PM
> |To: DISPATCH list
> |Subject: Re: [dispatch] Final Charter for Session-ID?
> |
> |I have a problem with this proposed charter.
> |
> |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
> |
> |>
> |> The definition of a higher-layer "session" of the same "message-flow"
> is
> |not defined by SIP when it involves a B2BUA, and is difficult to
> normatively
> |define in practice.  For the purposes of this WG charter's scope, two
> or
> |more dialogs of a B2BUA are correlated under any conditions where
> |correlation might aid troubleshooting, by identifying them as belonging
> to
> |the same higher-level session. Any solution documents from this working
> |group may expand or constrain the scope of their mechanism's
> correlation, if
> |the working group so decides.
> |
> |I think the above text is way too vague and impossible to reduce to
> |engineering practice. If you have 10 calls going to a PSTN GWs that is
> |having voice quality problems, clearly having all of them grouped
> together
> |is useful in debugging. Yet I doubt we want all the calls from a given
> PSTN
> |GW to have the same correlation ID. Our lack of any solid definition of
> what
> |it is we are trying to correlate is a strong indication to me that we
> have
> |not yet come to any consensus about what the problem is this WG would
> be
> |trying to fix.
> |
> |I would like to see a WG charted in this area but I would expect the
> IESG to
> |more or less laugh at a problem statement that reduced down to "uh
> might be
> |useful for debuggin". Can some people please try and propose a more
> concrete
> |definition of when two sip dialogs will be correlated?
> |
> |Cullen
> |
> |_______________________________________________
> |dispatch mailing list
> |dispatch@ietf.org
> |https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From christer.holmberg@ericsson.com  Thu Jun 10 05:57:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Muthu ArulMozhi Perumal (mperumal)" <mperumal@cisco.com>
Date: Thu, 10 Jun 2010 14:57:44 +0200
Thread-Topic: [dispatch] Final Charter for Session-ID?
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Hi,

Isn't the linkage the fact that C is the person A addressed for this specif=
ic session?

Regards,

Christer

=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 10. kes=E4kuuta 2010 15:54
> To: Muthu ArulMozhi Perumal (mperumal)
> Cc: DISPATCH list
> Subject: Re: [dispatch] Final Charter for Session-ID?
>=20
>=20
>=20
> Muthu ArulMozhi Perumal (mperumal) wrote:
> > Here is an initial attempt:
> >=20
> > If there is a dialog between A and B then A and B are in a session.
> > If there is a dialog between A and B and a dialog between B=20
> and C at=20
> > the same time then A, B and C are in a session.
> > More generally, if two or more participants are part of a session S=20
> > and at the same time there is a dialog between one of the=20
> participants=20
> > of S and another participant Z who is not part of S, then S=20
> (union) Z=20
> > is a session.
>=20
> The above just reiterates the problem that Cullen was raising.
> Just because B has a dialog with A and a dialog with C does=20
> not mean that there is a session between A and C. A gateway=20
> provides a perfect example of that.
>=20
> There needs to be some semantic link between the dialogs. The=20
> question is defining what sorts of semantic links should be=20
> used to share a session id.
>=20
> 	Thanks,
> 	Paul
>=20
> > I think this would cover most of SBC, 3PCC an app server=20
> cases. People=20
> > can add cases like park/retrieve if they think they should be=20
> > considered as part of a single session.
> >=20
> > Muthu
> >=20
> > |-----Original Message-----
> > |From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On
> > Behalf
> > |Of Cullen Jennings (fluffy)
> > |Sent: Wednesday, June 09, 2010 8:23 PM
> > |To: DISPATCH list
> > |Subject: Re: [dispatch] Final Charter for Session-ID?
> > |
> > |I have a problem with this proposed charter.
> > |
> > |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
> > |
> > |>
> > |> The definition of a higher-layer "session" of the same=20
> "message-flow"
> > is
> > |not defined by SIP when it involves a B2BUA, and is difficult to
> > normatively
> > |define in practice.  For the purposes of this WG charter's=20
> scope, two
> > or
> > |more dialogs of a B2BUA are correlated under any conditions where=20
> > |correlation might aid troubleshooting, by identifying them as=20
> > |belonging
> > to
> > |the same higher-level session. Any solution documents from this=20
> > |working group may expand or constrain the scope of their=20
> mechanism's
> > correlation, if
> > |the working group so decides.
> > |
> > |I think the above text is way too vague and impossible to=20
> reduce to=20
> > |engineering practice. If you have 10 calls going to a PSTN=20
> GWs that=20
> > |is having voice quality problems, clearly having all of=20
> them grouped
> > together
> > |is useful in debugging. Yet I doubt we want all the calls from a=20
> > |given
> > PSTN
> > |GW to have the same correlation ID. Our lack of any solid=20
> definition=20
> > |of
> > what
> > |it is we are trying to correlate is a strong indication to=20
> me that we
> > have
> > |not yet come to any consensus about what the problem is=20
> this WG would
> > be
> > |trying to fix.
> > |
> > |I would like to see a WG charted in this area but I would=20
> expect the
> > IESG to
> > |more or less laugh at a problem statement that reduced down to "uh
> > might be
> > |useful for debuggin". Can some people please try and propose a more
> > concrete
> > |definition of when two sip dialogs will be correlated?
> > |
> > |Cullen
> > |
> > |_______________________________________________
> > |dispatch mailing list
> > |dispatch@ietf.org
> > |https://www.ietf.org/mailman/listinfo/dispatch
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From mperumal@cisco.com  Thu Jun 10 06:22:51 2010
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From: "Muthu ArulMozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Yes, the fact that A and C are communicating is the linkage -- B is just =
facilitating that communication.

thanks,
Muthu=20

|-----Original Message-----
|From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
|Sent: Thursday, June 10, 2010 6:28 PM
|To: Paul Kyzivat (pkyzivat); Muthu ArulMozhi Perumal (mperumal)
|Cc: DISPATCH list
|Subject: RE: [dispatch] Final Charter for Session-ID?
|
|
|Hi,
|
|Isn't the linkage the fact that C is the person A addressed for this
|specific session?
|
|Regards,
|
|Christer
|
|> -----Original Message-----
|> From: dispatch-bounces@ietf.org
|> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
|> Sent: 10. kes=E4kuuta 2010 15:54
|> To: Muthu ArulMozhi Perumal (mperumal)
|> Cc: DISPATCH list
|> Subject: Re: [dispatch] Final Charter for Session-ID?
|>
|>
|>
|> Muthu ArulMozhi Perumal (mperumal) wrote:
|> > Here is an initial attempt:
|> >
|> > If there is a dialog between A and B then A and B are in a session.
|> > If there is a dialog between A and B and a dialog between B
|> and C at
|> > the same time then A, B and C are in a session.
|> > More generally, if two or more participants are part of a session S
|> > and at the same time there is a dialog between one of the
|> participants
|> > of S and another participant Z who is not part of S, then S
|> (union) Z
|> > is a session.
|>
|> The above just reiterates the problem that Cullen was raising.
|> Just because B has a dialog with A and a dialog with C does
|> not mean that there is a session between A and C. A gateway
|> provides a perfect example of that.
|>
|> There needs to be some semantic link between the dialogs. The
|> question is defining what sorts of semantic links should be
|> used to share a session id.
|>
|> 	Thanks,
|> 	Paul
|>
|> > I think this would cover most of SBC, 3PCC an app server
|> cases. People
|> > can add cases like park/retrieve if they think they should be
|> > considered as part of a single session.
|> >
|> > Muthu
|> >
|> > |-----Original Message-----
|> > |From: dispatch-bounces@ietf.org
|> [mailto:dispatch-bounces@ietf.org] On
|> > Behalf
|> > |Of Cullen Jennings (fluffy)
|> > |Sent: Wednesday, June 09, 2010 8:23 PM
|> > |To: DISPATCH list
|> > |Subject: Re: [dispatch] Final Charter for Session-ID?
|> > |
|> > |I have a problem with this proposed charter.
|> > |
|> > |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
|> > |
|> > |>
|> > |> The definition of a higher-layer "session" of the same
|> "message-flow"
|> > is
|> > |not defined by SIP when it involves a B2BUA, and is difficult to
|> > normatively
|> > |define in practice.  For the purposes of this WG charter's
|> scope, two
|> > or
|> > |more dialogs of a B2BUA are correlated under any conditions where
|> > |correlation might aid troubleshooting, by identifying them as
|> > |belonging
|> > to
|> > |the same higher-level session. Any solution documents from this
|> > |working group may expand or constrain the scope of their
|> mechanism's
|> > correlation, if
|> > |the working group so decides.
|> > |
|> > |I think the above text is way too vague and impossible to
|> reduce to
|> > |engineering practice. If you have 10 calls going to a PSTN
|> GWs that
|> > |is having voice quality problems, clearly having all of
|> them grouped
|> > together
|> > |is useful in debugging. Yet I doubt we want all the calls from a
|> > |given
|> > PSTN
|> > |GW to have the same correlation ID. Our lack of any solid
|> definition
|> > |of
|> > what
|> > |it is we are trying to correlate is a strong indication to
|> me that we
|> > have
|> > |not yet come to any consensus about what the problem is
|> this WG would
|> > be
|> > |trying to fix.
|> > |
|> > |I would like to see a WG charted in this area but I would
|> expect the
|> > IESG to
|> > |more or less laugh at a problem statement that reduced down to "uh
|> > might be
|> > |useful for debuggin". Can some people please try and propose a =
more
|> > concrete
|> > |definition of when two sip dialogs will be correlated?
|> > |
|> > |Cullen
|> > |
|> > |_______________________________________________
|> > |dispatch mailing list
|> > |dispatch@ietf.org
|> > |https://www.ietf.org/mailman/listinfo/dispatch
|> > _______________________________________________
|> > dispatch mailing list
|> > dispatch@ietf.org
|> > https://www.ietf.org/mailman/listinfo/dispatch
|> >
|> _______________________________________________
|> dispatch mailing list
|> dispatch@ietf.org
|> https://www.ietf.org/mailman/listinfo/dispatch
|>

From pkyzivat@cisco.com  Thu Jun 10 07:14:44 2010
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Muthu ArulMozhi Perumal (mperumal) wrote:
> Yes, the fact that A and C are communicating is the linkage -- B is just facilitating that communication.

Then that needs to be specified.

	Thanks,
	Paul


> thanks,
> Muthu 
> 
> |-----Original Message-----
> |From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> |Sent: Thursday, June 10, 2010 6:28 PM
> |To: Paul Kyzivat (pkyzivat); Muthu ArulMozhi Perumal (mperumal)
> |Cc: DISPATCH list
> |Subject: RE: [dispatch] Final Charter for Session-ID?
> |
> |
> |Hi,
> |
> |Isn't the linkage the fact that C is the person A addressed for this
> |specific session?
> |
> |Regards,
> |
> |Christer
> |
> |> -----Original Message-----
> |> From: dispatch-bounces@ietf.org
> |> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> |> Sent: 10. kesäkuuta 2010 15:54
> |> To: Muthu ArulMozhi Perumal (mperumal)
> |> Cc: DISPATCH list
> |> Subject: Re: [dispatch] Final Charter for Session-ID?
> |>
> |>
> |>
> |> Muthu ArulMozhi Perumal (mperumal) wrote:
> |> > Here is an initial attempt:
> |> >
> |> > If there is a dialog between A and B then A and B are in a session.
> |> > If there is a dialog between A and B and a dialog between B
> |> and C at
> |> > the same time then A, B and C are in a session.
> |> > More generally, if two or more participants are part of a session S
> |> > and at the same time there is a dialog between one of the
> |> participants
> |> > of S and another participant Z who is not part of S, then S
> |> (union) Z
> |> > is a session.
> |>
> |> The above just reiterates the problem that Cullen was raising.
> |> Just because B has a dialog with A and a dialog with C does
> |> not mean that there is a session between A and C. A gateway
> |> provides a perfect example of that.
> |>
> |> There needs to be some semantic link between the dialogs. The
> |> question is defining what sorts of semantic links should be
> |> used to share a session id.
> |>
> |> 	Thanks,
> |> 	Paul
> |>
> |> > I think this would cover most of SBC, 3PCC an app server
> |> cases. People
> |> > can add cases like park/retrieve if they think they should be
> |> > considered as part of a single session.
> |> >
> |> > Muthu
> |> >
> |> > |-----Original Message-----
> |> > |From: dispatch-bounces@ietf.org
> |> [mailto:dispatch-bounces@ietf.org] On
> |> > Behalf
> |> > |Of Cullen Jennings (fluffy)
> |> > |Sent: Wednesday, June 09, 2010 8:23 PM
> |> > |To: DISPATCH list
> |> > |Subject: Re: [dispatch] Final Charter for Session-ID?
> |> > |
> |> > |I have a problem with this proposed charter.
> |> > |
> |> > |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
> |> > |
> |> > |>
> |> > |> The definition of a higher-layer "session" of the same
> |> "message-flow"
> |> > is
> |> > |not defined by SIP when it involves a B2BUA, and is difficult to
> |> > normatively
> |> > |define in practice.  For the purposes of this WG charter's
> |> scope, two
> |> > or
> |> > |more dialogs of a B2BUA are correlated under any conditions where
> |> > |correlation might aid troubleshooting, by identifying them as
> |> > |belonging
> |> > to
> |> > |the same higher-level session. Any solution documents from this
> |> > |working group may expand or constrain the scope of their
> |> mechanism's
> |> > correlation, if
> |> > |the working group so decides.
> |> > |
> |> > |I think the above text is way too vague and impossible to
> |> reduce to
> |> > |engineering practice. If you have 10 calls going to a PSTN
> |> GWs that
> |> > |is having voice quality problems, clearly having all of
> |> them grouped
> |> > together
> |> > |is useful in debugging. Yet I doubt we want all the calls from a
> |> > |given
> |> > PSTN
> |> > |GW to have the same correlation ID. Our lack of any solid
> |> definition
> |> > |of
> |> > what
> |> > |it is we are trying to correlate is a strong indication to
> |> me that we
> |> > have
> |> > |not yet come to any consensus about what the problem is
> |> this WG would
> |> > be
> |> > |trying to fix.
> |> > |
> |> > |I would like to see a WG charted in this area but I would
> |> expect the
> |> > IESG to
> |> > |more or less laugh at a problem statement that reduced down to "uh
> |> > might be
> |> > |useful for debuggin". Can some people please try and propose a more
> |> > concrete
> |> > |definition of when two sip dialogs will be correlated?
> |> > |
> |> > |Cullen
> |> > |
> |> > |_______________________________________________
> |> > |dispatch mailing list
> |> > |dispatch@ietf.org
> |> > |https://www.ietf.org/mailman/listinfo/dispatch
> |> > _______________________________________________
> |> > dispatch mailing list
> |> > dispatch@ietf.org
> |> > https://www.ietf.org/mailman/listinfo/dispatch
> |> >
> |> _______________________________________________
> |> dispatch mailing list
> |> dispatch@ietf.org
> |> https://www.ietf.org/mailman/listinfo/dispatch
> |>
> 

From christer.holmberg@ericsson.com  Fri Jun 11 04:03:19 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: DISPATCH list <dispatch@ietf.org>
Date: Fri, 11 Jun 2010 13:03:15 +0200
Thread-Topic: RFC 3680 extensibility
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Hi,

Section 5.1 of RFC 3680 (reg event package) says:

"Other elements from different namespaces MAY be present for the
purposes of extensibility; elements or attributes from unknown
namespaces MUST be ignored."

However, there is no policy regarding whether the definition of these "othe=
r namespaces" needs to be done in IETF - only that they must be ignored if =
not supported.

3GPP has been discussing a couple of IMS specific cases, in which such exte=
nsion elements/attributes could be used as solution (no decissions have bee=
n made yet), so the question is whether it would require IETF work.

Regards,

Christer









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<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Section 5.1 of RFC 3680 (reg event package) says:</fo=
nt></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&quot;Other elements from different namespaces MAY be=
 present for the</font></div>
<div><font size=3D"2">purposes of extensibility; elements or attributes fro=
m unknown</font></div>
<div><font size=3D"2">namespaces MUST be ignored.&quot;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">However, there is no policy regarding whether the def=
inition of these &quot;other namespaces&quot; needs to be done in IETF - on=
ly that they must be ignored if not supported.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">3GPP has been discussing a couple of IMS specific cas=
es, in which such extension elements/attributes could be used as solution (=
no decissions have been made yet), so the question is whether it would requ=
ire IETF work.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
</font>
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From gonzalo.camarillo@ericsson.com  Fri Jun 11 04:59:02 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi,

yes, Mary's description is correct. The process is as follows. Once we
agreed on the charter, I requested the IESG to review it. This charter
will be discussed in the next IESG telechat (the coming Thursday),
actually. Once the IESG is happy with it, the next step will be to have
an IETF-wide review. After that, we will be able to charter the WG.

In any case, as we have said a number of times, proponents of a
particular piece of work should not stop working on it while we charter
their WG. They are encouraged to continue making progress in parallel
with the chartering process.

Cheers,

Gonzalo


On 09/06/2010 8:13 PM, Mary Barnes wrote:
> Hi Laura,
> 
> We did not consider this topic for pre-IETF-78 focus for the DISPATCH
> WG because the ADs are already in the process of preparing the charter
> for approval by the IESG, as I understood Gonzalo's emails on the
> topic.   I'll let ADs discuss the timeline for the chartering and
> really it would be up to them as to whether a meeting agenda slot
> would be requested for the topic.
> 
> The current model we've been using is that past topics that have been
> agreed to be chartered have already been "dispatched" per se, per the
> summaries in the wiki:
> http://trac.tools.ietf.org/wg/dispatch/trac/wiki
> 
> Thanks,
> Mary.
> 
> 
> 
> 
> On Wed, Jun 9, 2010 at 2:19 AM, Laura Liess
> <laura.liess.dt@googlemail.com> wrote:
>> Hi Mary, Gonzalo,
>>
>> What about the Alert-Info URN?  We had a lot of discussion about the
>> charter and name, my opinion was that we had consensus on the charter.
>> Dale would chair the WG.  I intend to submit a new draft version by
>> the end of June.
>> I also sent to you an e-mail some time ago asking for a time slot for
>> discussion in Masstricht. Is it anything I missed to do to get agenda
>> time?
>>
>> Thanks a lot
>> Laura
>>
>>
>>
>>
>> 2010/6/8 Mary Barnes <mary.ietf.barnes@gmail.com>:
>>> Hi all,
>>>
>>> Per the reminder posted on the ML on May 4th and as listed on the
>>> DISPATCH WG wiki, charter proposals for the topics to be
>>> handled/dispatched prior to and at IETF-78 were due on May 31st.
>>>
>>> We received charter proposals (problem statements/deliverables) for the
>>> following, with links to most recent charter proposals and discussion threads.
>>>
>>> o Session-ID:  updated charter proposal available
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01967.html
>>>
>>> o Telepresence:  updated charter proposal available
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01936.html
>>>
>>> o VIPR:  charter proposal
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01930.html
>>>
>>> o TEL URI WG proposal:  discussion to focus on problem statement and scope
>>> http://www.ietf.org/mail-archive/web/dispatch/current/msg01920.html
>>>
>>> The above items are the current targets for IETF-78 discussions.  Based
>>> on those discussions, agenda time will be allocated, items dispatched
>>> and adhocs scheduled as appropriate. Note, that the minimum time we
>>> would allocate to a topic is 30 minutes and some may warrant 45
>>> minutes. If we schedule adhocs, we will try to have those announced
>>> around the time the agenda is finalized.
>>>
>>> The following items have also been discussed on the mailing list, but
>>> are not yet ready for dispatching:
>>>
>>> o NGN Reason: Some interest. Needs more discussion and scoping.
>>> http://datatracker.ietf.org/doc/draft-jesske-dispatch-reason-in-responses/
>>> http://datatracker.ietf.org/doc/draft-jesske-dispatch-req-reason-in-responses/
>>>
>>> o PAN - wait for more energy
>>> http://datatracker.ietf.org/doc/draft-lawrence-sipforum-provider-alias/
>>>
>>> o  ACH - ACH analysis draft to happen as part of BLISS wind down
>>> http://datatracker.ietf.org/doc/draft-ietf-bliss-ach-analysis/
>>>
>>> o  ACH - HTTP API docs - need some hallway discussions
>>> http://datatracker.ietf.org/doc/draft-yusef-dispatch-ach-rest-api/
>>>
>>> o  Media security - needs additional ML discussion
>>> http://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-parameter/
>>>
>>> o Communication diversion and barring notification - feedback thus far
>>> needs to be considered - i.e, problem statement to be clarified and
>>> alternate solutions to be consider (Note: IPR on the communication
>>> diversion draft)
>>> http://datatracker.ietf.org/doc/draft-avasarala-dispatch-comm-div-notification/
>>> http://datatracker.ietf.org/doc/-avasarala-dispatch-comm-barring-notification/
>>>
>>>
>>> As a reminder, the following are the cutoffs for drafts, so please make
>>> sure that any drafts relevant to the topic are submitted prior to those
>>> deadlines:
>>> * - 00 documents:  July 5th (< 4 weeks)
>>> * all other documents: July 12th (< 5 weeks)
>>>
>>> Please keep in mind that the focus of discussions should be the problem
>>> statement, scope and proposed deliverables for the topic.
>>>
>>> As a reminder, items can be dispatched without being discussed at a f2f
>>> meeting and the most effective way to achieve this is to have problem
>>> statements, scope of topics and any relevant documents available for
>>> discussion, rather than documents focusing on solutions.
>>>
>>> Please, let the chairs know if there are any concerns.
>>>
>>> Thanks,
>>> Mary Barnes and Cullen Jennings
>>> DISPATCH WG chairs
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>


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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, DISPATCH list <dispatch@ietf.org>
Date: Fri, 11 Jun 2010 12:06:29 -0400
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Ch=
rister Holmberg [christer.holmberg@ericsson.com]

Section 5.1 of RFC 3680 (reg event package) says:

"Other elements from different namespaces MAY be present for the
purposes of extensibility; elements or attributes from unknown
namespaces MUST be ignored."

However, there is no policy regarding whether the definition of these "othe=
r namespaces" needs to be done in IETF - only that they must be ignored if =
not supported.

3GPP has been discussing a couple of IMS specific cases, in which such exte=
nsion elements/attributes could be used as solution (no decissions have bee=
n made yet), so the question is whether it would require IETF work.
________________________________________

The rule that unknown elements/attributes must be ignored covers you there,=
 doesn't it?  If a non-3GPP system subscribes to an "extended" event packag=
e, it will ignore the extensions.

Dale

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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Sun, 13 Jun 2010 22:37:12 -0400
Thread-Topic: [dispatch] DISPATCH-78 topics - Alert URNs
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________________________________________
From: Gonzalo Camarillo [Gonzalo.Camarillo@ericsson.com]

yes, Mary's description is correct. The process is as follows. Once we
agreed on the charter, I requested the IESG to review it. This charter
will be discussed in the next IESG telechat (the coming Thursday),
actually. Once the IESG is happy with it, the next step will be to have
an IETF-wide review. After that, we will be able to charter the WG.

In any case, as we have said a number of times, proponents of a
particular piece of work should not stop working on it while we charter
their WG. They are encouraged to continue making progress in parallel
with the chartering process.
________________________________________

That is all fine, but what are the implications regarding getting a time sl=
ot at Maastricht?  There seem to be two choices:  a slot within Dispatch, o=
r a separately scheduled Salud session.  What we do not want to get stuck w=
ith is not being able to get time in Dispatch (because a charter has been p=
roposed) and not being able to get a separate session (because the charter =
hasn't gone through all the approval steps), as that would prevent any sess=
ion at  Maastricht.

Dale

From gonzalo.camarillo@ericsson.com  Sun Jun 13 22:55:58 2010
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Date: Mon, 14 Jun 2010 08:55:57 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: DISPATCH <dispatch@ietf.org>, Robert Sparks <RjS@nostrum.com>
Subject: Re: [dispatch] DISPATCH-78 topics - Alert URNs
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Hi,

if you need face-to-face time for this, go ahead talk to the DISPATCH
chairs so that you can either get a slot within the DISPATCH session or,
alternatively, have an ad-hoc meeting. If we managed to charter the WG
in time, we would have a WG meeting instead.

Cheers,

Gonzalo

On 14/06/2010 5:37 AM, WORLEY, Dale R (Dale) wrote:
> ________________________________________
> From: Gonzalo Camarillo [Gonzalo.Camarillo@ericsson.com]
> 
> yes, Mary's description is correct. The process is as follows. Once we
> agreed on the charter, I requested the IESG to review it. This charter
> will be discussed in the next IESG telechat (the coming Thursday),
> actually. Once the IESG is happy with it, the next step will be to have
> an IETF-wide review. After that, we will be able to charter the WG.
> 
> In any case, as we have said a number of times, proponents of a
> particular piece of work should not stop working on it while we charter
> their WG. They are encouraged to continue making progress in parallel
> with the chartering process.
> ________________________________________
> 
> That is all fine, but what are the implications regarding getting a time slot at Maastricht?  There seem to be two choices:  a slot within Dispatch, or a separately scheduled Salud session.  What we do not want to get stuck with is not being able to get time in Dispatch (because a charter has been proposed) and not being able to get a separate session (because the charter hasn't gone through all the approval steps), as that would prevent any session at  Maastricht.
> 
> Dale


From laura.liess.dt@googlemail.com  Sun Jun 13 23:14:18 2010
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From: Laura Liess <laura.liess.dt@googlemail.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,  "WORLEY, Dale R (Dale)" <dworley@avaya.com>, Mary Barnes <mary.ietf.barnes@gmail.com>
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Hi,

Dale is right. We need a face2face meeting again.

Thanks,
Laura

2010/6/14 Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>:
> Hi,
>
> if you need face-to-face time for this, go ahead talk to the DISPATCH
> chairs so that you can either get a slot within the DISPATCH session or,
> alternatively, have an ad-hoc meeting. If we managed to charter the WG
> in time, we would have a WG meeting instead.
>
> Cheers,
>
> Gonzalo
>
> On 14/06/2010 5:37 AM, WORLEY, Dale R (Dale) wrote:
>> ________________________________________
>> From: Gonzalo Camarillo [Gonzalo.Camarillo@ericsson.com]
>>
>> yes, Mary's description is correct. The process is as follows. Once we
>> agreed on the charter, I requested the IESG to review it. This charter
>> will be discussed in the next IESG telechat (the coming Thursday),
>> actually. Once the IESG is happy with it, the next step will be to have
>> an IETF-wide review. After that, we will be able to charter the WG.
>>
>> In any case, as we have said a number of times, proponents of a
>> particular piece of work should not stop working on it while we charter
>> their WG. They are encouraged to continue making progress in parallel
>> with the chartering process.
>> ________________________________________
>>
>> That is all fine, but what are the implications regarding getting a time=
 slot at Maastricht? =A0There seem to be two choices: =A0a slot within Disp=
atch, or a separately scheduled Salud session. =A0What we do not want to ge=
t stuck with is not being able to get time in Dispatch (because a charter h=
as been proposed) and not being able to get a separate session (because the=
 charter hasn't gone through all the approval steps), as that would prevent=
 any session at =A0Maastricht.
>>
>> Dale
>
>

From gonzalo.camarillo@ericsson.com  Sun Jun 13 23:42:20 2010
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Subject: Re: [dispatch] RFC 3680 extensibility
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Hi,

yes, I agree with Dale that, technically, the system is ready to handle
interoperability even in presence of unknown stuff... but the question
here is whether or not we have a policy that regulates the creation of
such extensions.

XML namespaces are not covered by RFC 5727. However, Section 4.2 of RFC
3863 specifies rules for registering extensions to the basic presence
information data format:

http://tools.ietf.org/html/rfc3863#section-4.2

Considering the spirit of RFC 5727 and the example above, a potential
policy could be to register reg-event extensions with the IANA using the
Specification Required policy defined in RFC 5226. In this case, this
would mean that the extension would be defined in a 3GPP spec and
revised by an expert who would be designated by the RAI ADs. The expert
reviewer would make sure that new extensions do not overlap with current
IETF work.

In any case, we would welcome further comments on this issue.

Thanks,

Gonzalo


On 11/06/2010 7:06 PM, WORLEY, Dale R (Dale) wrote:
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Christer Holmberg [christer.holmberg@ericsson.com]
> 
> Section 5.1 of RFC 3680 (reg event package) says:
> 
> "Other elements from different namespaces MAY be present for the
> purposes of extensibility; elements or attributes from unknown
> namespaces MUST be ignored."
> 
> However, there is no policy regarding whether the definition of these "other namespaces" needs to be done in IETF - only that they must be ignored if not supported.
> 
> 3GPP has been discussing a couple of IMS specific cases, in which such extension elements/attributes could be used as solution (no decissions have been made yet), so the question is whether it would require IETF work.
> ________________________________________
> 
> The rule that unknown elements/attributes must be ignored covers you there, doesn't it?  If a non-3GPP system subscribes to an "extended" event package, it will ignore the extensions.
> 
> Dale
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


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To: <Gonzalo.Camarillo@ericsson.com>, <dworley@avaya.com>
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Gonzalo 

I would pretty much agree with this approach. Only thing I would add is that=
 the expert reviewer should also verify that there is no implied Require opt=
ion-tag i.e. that subscribers supporting the extension still can perform bas=
ic functionality with notifiers that do not support the extension otherwise=
 I think we require new option-tags and that means standards track. 

Andrew

----- Original Message -----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
Sent: Monday, June 14, 2010 02:42 AM
To: WORLEY, Dale R (Dale) <dworley@avaya.com>
Cc: DISPATCH list <dispatch@ietf.org>
Subject: Re: [dispatch] RFC 3680 extensibility

Hi,

yes, I agree with Dale that, technically, the system is ready to handle
interoperability even in presence of unknown stuff... but the question
here is whether or not we have a policy that regulates the creation of
such extensions.

XML namespaces are not covered by RFC 5727. However, Section 4.2 of RFC
3863 specifies rules for registering extensions to the basic presence
information data format:

http://tools.ietf.org/html/rfc3863#section-4.2

Considering the spirit of RFC 5727 and the example above, a potential
policy could be to register reg-event extensions with the IANA using the
Specification Required policy defined in RFC 5226. In this case, this
would mean that the extension would be defined in a 3GPP spec and
revised by an expert who would be designated by the RAI ADs. The expert
reviewer would make sure that new extensions do not overlap with current
IETF work.

In any case, we would welcome further comments on this issue.

Thanks,

Gonzalo


On 11/06/2010 7:06 PM, WORLEY, Dale R (Dale) wrote:
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of C=
hrister Holmberg [christer.holmberg@ericsson.com]
> 
> Section 5.1 of RFC 3680 (reg event package) says:
> 
> "Other elements from different namespaces MAY be present for the
> purposes of extensibility; elements or attributes from unknown
> namespaces MUST be ignored."
> 
> However, there is no policy regarding whether the definition of these "oth=
er namespaces" needs to be done in IETF - only that they must be ignored if=
 not supported.
> 
> 3GPP has been discussing a couple of IMS specific cases, in which such ext=
ension elements/attributes could be used as solution (no decissions have bee=
n made yet), so the question is whether it would require IETF work.
> ________________________________________
> 
> The rule that unknown elements/attributes must be ignored covers you there=
, doesn't it?  If a non-3GPP system subscribes to an "extended" event packag=
e, it will ignore the extensions.
> 
> Dale
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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From christer.holmberg@ericsson.com  Mon Jun 14 02:17:54 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>, "WORLEY, Dale R (Dale)" <dworley@avaya.com>
Date: Mon, 14 Jun 2010 11:17:28 +0200
Thread-Topic: [dispatch] RFC 3680 extensibility
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Hi,

So, it would be the same policy as we have for info packages.

Thanks!

Regards,

Christer

=20

> -----Original Message-----
> From: Gonzalo Camarillo=20
> Sent: 14. kes=E4kuuta 2010 9:42
> To: WORLEY, Dale R (Dale)
> Cc: Christer Holmberg; DISPATCH list
> Subject: Re: [dispatch] RFC 3680 extensibility
>=20
> Hi,
>=20
> yes, I agree with Dale that, technically, the system is ready=20
> to handle interoperability even in presence of unknown=20
> stuff... but the question here is whether or not we have a=20
> policy that regulates the creation of such extensions.
>=20
> XML namespaces are not covered by RFC 5727. However, Section=20
> 4.2 of RFC
> 3863 specifies rules for registering extensions to the basic=20
> presence information data format:
>=20
> http://tools.ietf.org/html/rfc3863#section-4.2
>=20
> Considering the spirit of RFC 5727 and the example above, a=20
> potential policy could be to register reg-event extensions=20
> with the IANA using the Specification Required policy defined=20
> in RFC 5226. In this case, this would mean that the extension=20
> would be defined in a 3GPP spec and revised by an expert who=20
> would be designated by the RAI ADs. The expert reviewer would=20
> make sure that new extensions do not overlap with current IETF work.
>=20
> In any case, we would welcome further comments on this issue.
>=20
> Thanks,
>=20
> Gonzalo
>=20
>=20
> On 11/06/2010 7:06 PM, WORLEY, Dale R (Dale) wrote:
> > ________________________________________
> > From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org]=20
> On Behalf=20
> > Of Christer Holmberg [christer.holmberg@ericsson.com]
> >=20
> > Section 5.1 of RFC 3680 (reg event package) says:
> >=20
> > "Other elements from different namespaces MAY be present for the=20
> > purposes of extensibility; elements or attributes from unknown=20
> > namespaces MUST be ignored."
> >=20
> > However, there is no policy regarding whether the=20
> definition of these "other namespaces" needs to be done in=20
> IETF - only that they must be ignored if not supported.
> >=20
> > 3GPP has been discussing a couple of IMS specific cases, in=20
> which such extension elements/attributes could be used as=20
> solution (no decissions have been made yet), so the question=20
> is whether it would require IETF work.
> > ________________________________________
> >=20
> > The rule that unknown elements/attributes must be ignored=20
> covers you there, doesn't it?  If a non-3GPP system=20
> subscribes to an "extended" event package, it will ignore the=20
> extensions.
> >=20
> > Dale
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20
> =

From dworley@avaya.com  Mon Jun 14 18:26:33 2010
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Talking with Laura Liess, it looks like the "Alert-Info URN" effort will ne=
ed a time slot of 30 to 45 minutes at Maastricht.  (We have only one draft =
in progress.)  There is ambiguity regarding where in the schedule this shou=
ld be put, as the working group is just being approved.  However, given the=
 relatively small amount of time needed, it seems best to me if it could be=
 put at the end of a Dispatch session or one of the other SIP sessions.  I =
am told that we already have a time request submitted, but we have heard no=
 response regarding it.

What is likely to work best for the Maastricht meeting?

Dale

From Peter.Dawes@vodafone.com  Tue Jun 15 04:37:44 2010
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Hi,
I have submitted a revised draft on media plane security on the first
hop to include more background on motivation, add SDP to the examples,
and indicate that the entity on the edge of the network is a B2BUA not a
proxy.=20

https://datatracker.ietf.org/doc/draft-dawes-dispatch-mediasec-parameter
/

I have copied the revised motivations clause below.=20

Regards,
Peter


1.1.  Motivations

1.1.1.  Access Network Protection

   Some access technologies protect the data passed over them by
   default, for example many cellular wireless accesses, but some do
   not, for example WLAN.  For accesses with no protection, it is useful
   for the media controlled by SIP signalling to be protected by default
   because of vulnerability to eavesdropping.  It is currently possible
   for a UA to request protection of the media plane end-to-end by
   including the crypto attribute in SDP at session setup.  This does
   not guarantee protection however, because it relies on support of
   encryption by the called UA, or by another entity in the path taken
   by the media.  In some cases, the session will originate in an access
   that protects the media and terminate in one that does not, meaning
   that media is protected in all but some hops of its path.  In cases
   where the same provider supplies the user equipment and provides the
   IP access, the IP access technology that the UA will use is
   predictable and the media is vulnerable only as far as the core
   network.  In such cases, the user equipment it is possible to protect
   the media plane by encrypting at the UA and decrypting at the edge of
   the core network, and for the user agent that originates or
   terminates the session to expect the edge of the core network to be
   capable of encrypting and decrypting media.  This document describes
   this case of first-hop protection, which is typically provided by
   default to a user agent.  Both media and signalling must pass through
   the entity at the edge of the core network, which must therefore be a
   back-to-back user agent (B2BUA).

   End to access edge media protection described in this document is not
   a substitute for end-to-end media protection.  A user agent requests
   end-to-access-edge media protection by including a "a=3D3ge2ae" SDP
   attribute at session setup.  If this attribute is not included, then



Dawes                   Expires December 17, 2010               [Page 4]
=0C
Internet-Draft          Media Security Parameter               June 2010


   end-to-end protection is expected by the user agent and protection
   MUST NOT fall back to end-to-access-edge protection.

1.1.2.  DTLS-SRTP

   DTLS-SRTP is described in RFC 5763 [8], which shows the basic message
   flow in Figure 1.  The DTLS handshake takes place when Bob receives
   the initial INVITE request.  Alice uses the source IP address of the
   DTLS "hello" message (3) as the destination address for "hello"
   message (4) even though Bob does not provide a contact address until
   the 200 OK message (8).



                           Alice            Proxies             Bob
                |(1) INVITE       |                  |
                |---------------->|                  |
                |                 |(2) INVITE        |
                |                 |----------------->|
                |                 |(3) hello         |
                |<-----------------------------------|
                |(4) hello        |                  |
                |----------------------------------->|
                |                 |(5) finished      |
                |<-----------------------------------|
                |                 |(6) media         |
                |<-----------------------------------|
                |(7) finished     |                  |
                |----------------------------------->|
                |                 |(8)  200 OK       |
                |                 |<-----------------|
                |(9)  200 OK      |                  |
                |<----------------|                  |
                |                 |(10) media        |
                |<---------------------------------->|
                |(11) ACK         |                  |
                |----------------------------------->|

      Message (1):  INVITE Alice -> Proxy


            Figure 1: Security capability exchange message flow

   In some networks, media packets are blocked between Alice and Bob
   until Alice receives the 200 (OK) message 9, which blocks the DTLS
   handshake until after Alice receives message (9).  Media will not be
   successfully exchanged unless the DTLS handshake is re-attempted
   after message (9) 200 OK.  Even if the handshake is re-attempted, the



Dawes                   Expires December 17, 2010               [Page 5]
=0C
Internet-Draft          Media Security Parameter               June 2010


   media will be clipped until the handshake is complete.

1.1.3.  SDP Capability Negotiation

   SDP capability negotiation is described in
   draft-ietf-mmusic-sdp-capability-negotiation [9] and describes
   sending multiple potential SDP combinations as an offer, such that a
   user agent can offer a choice of media security alternatives in the
   body of an initial INVITE request.  However, the caller UA has no
   prior knowledge of whether media plane security setup will succeed
   and in many cases it will fail or cause a lengthy delay while the
   user agent re-attempts, for example using a different IP access
   network.

1.1.4.  Motivations for RFC 3329

   RFC3329 describes why security is needed to protect SIP signalling
   from man-in-the-middle attacks, and to accomodate the expected wide
   variation in security mechanism support by SIP entities.  The media
   plane requires similar protection and exchange of security
   capabilities, for example to prevent eavesdropping in environments
   such as public wireless access networks that have no inherent
   security.  For the media plane security mechanism defined by this
   document, the cryptographic key is in plain text in SDP, therefore
   signalling SHOULD be protected e.g. using the security mechanism
   negotiation described by RFC 3329 [1]






From john.elwell@siemens-enterprise.com  Tue Jun 15 07:27:57 2010
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To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "dispatch@ietf.org" <dispatch@ietf.org>
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Thread-Topic: Reason In responses(draft-jesske-dispatch-reason-in-responses-02)
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=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: 07 June 2010 06:59
> To: Elwell, John; dispatch@ietf.org
> Subject: AW: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Hi John,
> Thank you for your comments:
> Here are my view.
>=20
> REQ-1:
> I think that REQ-1 is still a justification, because not each=20
> Operator will use SIP-T for tunnelling purposes.
> If you pass an untrusted domain SIP-T should not part,=20
> because it could include restricted numbers or other=20
> information that should not be presented to untrusted networks.
>=20
> Also only for passing the Q.850 Cause code to encapsulate the=20
> whole ISUP is a little over dimensioned.
> The SIP-T MIME can be lost due to the interdomain trust=20
> relation ships which the Reason header cannot.
>=20
> For in an multi carrier/service provider environment like=20
> Germany is, it will not be secure that each operator will=20
> support SIP-T.=20
[JRE] This helps to explain why SIP-T is not sufficient. IF SIP-T requires =
you to include additional information that you may not want to include when=
 going outside a domain, that would be a reason for requiring a different s=
olution. This was not made clear before.

John

>=20
> REQ-2
> It must be possible to provide an accurate indication to a=20
> UAC, so that the UAC can provide a suitable indication to the=20
> user.
>=20
> So that is the proposal for REQ-2.
>=20
> REQ-3=20
>    A UA may have the ability to display ISUP specific release=20
> causes or
>    show a equivalent text.
>=20
> So that would be sufficient enough for REQ-3. If people would=20
> like to see such text.
>=20
> REQ-3/4
> Your comment with regard to the type of UA is correct. I=20
> tried to satisfy people. The fact is that is not intended to=20
> include the Q.850 cause by an SIP end device. So the=20
> conclusion is changing REQ-3 and deleting REQ-4.
>=20
> So now I would like to ask the people which requirements they=20
> would like to see in the draft.
> As I understand John he sees only one (REQ-2)
>=20
> I propose REQ-1, -2 and -3.
>=20
> Opinions?
>=20
> Thank you and Best Regards
>=20
> Roland
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Zentrum Technik Einf=FChrung
> Roland Jesske
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628-2766 (Tel.)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobil)
> E-Mail: mailto:r.jesske@telekom.de
> http://www.telekom.com =20
>=20
> Erleben, was verbindet.=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> Albert Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> nicht jede E-Mail drucken.
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Gesendet: Dienstag, 18. Mai 2010 21:37
> An: Jesske, Roland; dispatch@ietf.org
> Betreff: RE: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Roland,=20
>=20
> > -----Original Message-----
> > From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> > Sent: 18 May 2010 15:57
> > To: Elwell, John; dispatch@ietf.org
> > Subject: AW: Reason In=20
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >=20
> > Hi John,
> > Thank you for your comments.=20
> > It was asked to split the draft into requirements and at=20
> > least procedures.
> >=20
> > You are right that within the requirements draft is more that=20
> > requirements. It is also the motivation, justification and=20
> > examples. I can change the title if people are happy with it.=20
> > Question is if such a draft is also needed as RFC or if it=20
> > only seen as a temporary document during the discussion.
> [JRE] I was not proposing a title change, and I have no=20
> problem combining motivation with requirements, although I do=20
> feel the examples look too much like solution and are=20
> unnecessary. Although some people asked you to split the=20
> draft, in my opinion it was wasted effort, but it is done now.
>=20
> >=20
> > With regard to REQ-1 I propose the following re-wording:
> >=20
> > It should be possible to support within native SIP=20
> > environment PSTN-SIP-PSTN scenarios where the reason of a=20
> > call release can be transferred though the SIP domain
> > without any loss of information and no change of reason.
> >=20
> > I think this was discussed lengthy during the last meeting=20
> > and email discussion.
> > Not each PSTN GW will support ISUP MIME encapsulation. At=20
> > least the 3GPP IMS has no SIP. The IMS is not using SIP-T.
> [JRE] I don't think the new wording makes any difference.=20
> SIP-T is a means for tunnelling a Q.850 cause through a=20
> native SIP domain. The IETF already defines SIP-T as the=20
> solution, and if this solution is not sufficient, you need to=20
> cite reasons why not and identify a requirement that cannot=20
> be met using SIP-T. Just saying that IMS does not use SIP-T=20
> doesn't sound like a sufficient reason to me, because clearly=20
> IMS could be made to use SIP-T.
>=20
> To be clear, I am not saying that the Reason header field=20
> should not be used in this situation, if the header field is=20
> defined for use in responses in general in order meet other=20
> requirements. I am simply saying that REQ-1 is not sufficient=20
> justification for allowing the Reason header field in=20
> responses, since REQ-1 can be satisfied using an existing mechanism.
>=20
>=20
> >=20
> > REQ-2 and REQ-3
> > Yes it looks similar.
> >=20
> > REQ-3 was added because it was seen from user device=20
> > perspective that the Device have the possibility to show the=20
> > Q.850 cause but that such a End Device does not include Q.850=20
> > causes. Of course a MGC should have this possibility.
> >=20
> > I would be happy to shrink REQ-2 and 3 to your proposed text.
> >=20
> > REQ-2
> > It must be possible to provide an accurate indication to a=20
> > UAC, so that the UAC can provide a suitable indication to the=20
> > user. Whether this be an announcement, a textual message, an=20
> > icon, a flashing lamp, a tone, a vibration, an odour or=20
> > whatever, is a user interface matter.
> [JRE] I did not intend the second sentence to be part of the=20
> requirement - it was just for illustration of why the=20
> existing text, which focused on textual display, was inadequate.
>=20
> >=20
> > REQ-4 was the consequence of REQ-3 where we do not allow the=20
> > inclusion of a Q850 cause by a end device.
> >=20
> > So the trick is not allow the SIP end device to include a=20
> > Q850 cause and in some certain cases to allow an service=20
> > provider controlled application server to include a Reason=20
> > header with a Q.850 cause. But it should not be done=20
> > generally by SIP applications. So normally a ISUP=20
> > implementation will use ISUP causes but not SIP implementations.
> > But there are SIP Application servers out in the world that=20
> > will have a ISUP application (or ISUP like application that=20
> > reacts and behaves as within a ISUP network).
> [JRE] I found REQ-3 rather confusing. On re-reading it, the=20
> first sentence is a requirement concerning reception, and the=20
> second sentence seems (if I understand correctly) to be a=20
> statement about sending. Assuming my understanding is=20
> correct, making a statement, within a requirement, that=20
> something is out of scope seems to suggest it is not part of=20
> the requirement.
>=20
> In addition, the way SIP is specified, there is no real=20
> distinction between different types of UA. In other words,=20
> SIP procedures for UAs apply to UAs in general, not to=20
> specific types of UA like ASs. So writing a requirement that=20
> tries to distinguish between different types of UA does not=20
> seem to make sense.
>=20
> >=20
> > Question is now how to proceed further.
> >=20
> > Are people happy to have the documents split or put again=20
> > together with striking out the examples ect?
> > Or to have the documents as they are with some transfer of=20
> > text to the draft-jesske-dispatch-reason-in-responses-02 draft?
> [JRE] I don't have a strong opinion about document structure,=20
> although my personal preference would have been to leave it=20
> as one document, as it was originally. My main concern is=20
> trying to come up with a meaningful set of requirements, and=20
> in my opinion it reduces to a single requirement - being able=20
> to convey a Q.850 cause from PSTN (or similar) to a UAC so=20
> that the UAC can render meaningful information to the user.
>=20
> John
>=20
>=20
> >=20
> > Comments?
> >=20
> > Thank you and Best Regards
> >=20
> > Roland
> >=20
> > -----Urspr=FCngliche Nachricht-----
> > Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> > Gesendet: Donnerstag, 13. Mai 2010 09:24
> > An: Jesske, Roland; dispatch@ietf.org
> > Betreff: RE: Reason In=20
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >=20
> > Roland,
> >=20
> > In my view, there is a simple requirement to convey Q.850=20
> > causes in SIP response messages, so that the UAC will receive=20
> > a more accurate indication of the reason for rejection when=20
> > rejection comes from PSTN. The cause mapping table helps to=20
> > illustrate this. This is not rocket science, and I am=20
> > disappointed that we have not found a quick way of=20
> > dispatching this and just getting the work done. In my=20
> > opinion the split into separate documents was unnecessary=20
> > (take as an example the recently-published RFC 5876, which=20
> > combines motivation and solution in one document), but it has=20
> > been done now. Unfortunately I don't think it has moved us=20
> > any further forward.
> >=20
> > I don't think the requirements in section 3 are particularly=20
> > well expressed. REQ-1 can already be met by tunnelling in=20
> > accordance with SIP-T. I would not expect the IETF to provide=20
> > a new solution specifically for that requirement.
> >=20
> > I think REQ-2 and REQ-3 boil down to the same thing - it must=20
> > be possible to provide an accurate indication to a UAC, so=20
> > that the UAC can provide a suitable indication to the user.=20
> > Whether this be an announcement, a textual message, an icon,=20
> > a flashing lamp, a tone, a vibration, an odour or whatever,=20
> > is a user interface matter.
> >=20
> > I also find REQ-4 rather problematic. If we have a solution=20
> > for PSTN, we would also have a solution for some other form=20
> > of gateway into a domain that provides a "PSTN-like service".=20
> > Unfortunately, if you try to bring this out as a separate=20
> > requirement, it raises the question of what exactly is this=20
> > "PSTN-like service", and if it is somehow SIP-based, why it=20
> > needs to be "PSTN-like", and so on.
> >=20
> > Basically I think there is only a single requirement here -=20
> > provision of a correct indication to a UAC when rejection=20
> > comes from the PSTN. I think that is a reasonable requirement=20
> > and we should just go ahead and do it, rather like we did=20
> > with RFC 5876 when we extended the range of messages that=20
> > PAI/PPI could be used in.
> >=20
> > I find the statement in REQ-3 "A inclusion of [Q.850] causes=20
> > is out of scope." very strange. I thought the whole idea was=20
> > to convey Q.850 causes, so why say they are out of scope? I=20
> > assume this is some sort of typo.
> >=20
> > I think one of the consequences of deciding to split=20
> > requirements and solution into separate drafts is that the=20
> > requirements draft has to stick to motivation and=20
> > requirements. I find too much in the draft (particularly the=20
> > examples) that smells of solution.
> >=20
> > John
> >=20
> >=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org=20
> > > [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> R.Jesske@telekom.de
> > > Sent: 13 May 2010 03:05
> > > To: dispatch@ietf.org
> > > Subject: Re: [dispatch] Reason In responses=20
> > > (draft-jesske-dispatch-reason-in-responses-02)
> > >=20
> > > Dear all,=20
> > > I have asked for comments on the two drafts mentioned below.=20
> > > So far I didn't get any. Does that mean that people are happy=20
> > > now with the content and we could proceed with it?=20
> > >=20
> > > Thank you and Best Regrads=20
> > >=20
> > > Roland=20
> > >=20
> > >=20
> > > _____________________________________________=20
> > > Von:    Jesske, Roland =20
> > > Gesendet:       Donnerstag, 8. April 2010 09:46=20
> > > An:     dispatch@ietf.org=20
> > > Betreff:        Reason In responses=20
> > > (draft-jesske-dispatch-reason-in-responses-02)=20
> > >=20
> > > Dear all,=20
> > > During the discussion concerning the Reason header field in=20
> > > Responses I was asked to split the document into an=20
> > > requirements part and an protocol part.
> > >=20
> > > So please feel free to comment on the drafts.=20
> > >=20
> > > http://64.170.98.42/html/draft-jesske-dispatch-reason-in-respo
> > > nses-02=20
> > > <http://64.170.98.42/html/draft-jesske-dispatch-reason-in-resp
> > > onses-02> =20
> > >=20
> > > http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-r
> > > esponses-00=20
> > > <http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-
> > > responses-00> =20
> > >=20
> > > Best regards=20
> > >=20
> > > Roland Jesske=20
> > >=20
> > >=20
> > > Deutsche Telekom Netzproduktion GmbH=20
> > > Zentrum Technik Einf=FChrung=20
> > > Roland Jesske=20
> > > Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
> > > +49 6151 628-2766 (Tel.)=20
> > > +49 521 9210-3753  (Fax)=20
> > > +49 171 8618-445  (Mobil)=20
> > > E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de> =20
> > > www.telekom.com <http:www.telekom.com>  =20
> > >=20
> > > Erleben, was verbindet.=20
> > >=20
> > > Deutsche Telekom Netzproduktion GmbH=20
> > > Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)=20
> > > Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> > > Albert Matheis, Klaus Peren=20
> > > Handelsregister: Amtsgericht Bonn HRB 14190=20
> > > Sitz der Gesellschaft: Bonn=20
> > > USt-IdNr. DE 814645262=20
> > >=20
> > > Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> > > nicht jede E-Mail drucken.=20
> > >=20
> > >=20
> > >=20
> >=20
> =

From fluffy@cisco.com  Tue Jun 15 07:28:49 2010
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I'm not following what you mean but it seems like this might lead to a =
workable definition. Can you propose something a bit more specific?


On Jun 10, 2010, at 6:57 AM, Christer Holmberg wrote:

>=20
> Hi,
>=20
> Isn't the linkage the fact that C is the person A addressed for this =
specific session?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org=20
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: 10. kes=E4kuuta 2010 15:54
>> To: Muthu ArulMozhi Perumal (mperumal)
>> Cc: DISPATCH list
>> Subject: Re: [dispatch] Final Charter for Session-ID?
>>=20
>>=20
>>=20
>> Muthu ArulMozhi Perumal (mperumal) wrote:
>>> Here is an initial attempt:
>>>=20
>>> If there is a dialog between A and B then A and B are in a session.
>>> If there is a dialog between A and B and a dialog between B=20
>> and C at=20
>>> the same time then A, B and C are in a session.
>>> More generally, if two or more participants are part of a session S=20=

>>> and at the same time there is a dialog between one of the=20
>> participants=20
>>> of S and another participant Z who is not part of S, then S=20
>> (union) Z=20
>>> is a session.
>>=20
>> The above just reiterates the problem that Cullen was raising.
>> Just because B has a dialog with A and a dialog with C does=20
>> not mean that there is a session between A and C. A gateway=20
>> provides a perfect example of that.
>>=20
>> There needs to be some semantic link between the dialogs. The=20
>> question is defining what sorts of semantic links should be=20
>> used to share a session id.
>>=20
>> 	Thanks,
>> 	Paul
>>=20
>>> I think this would cover most of SBC, 3PCC an app server=20
>> cases. People=20
>>> can add cases like park/retrieve if they think they should be=20
>>> considered as part of a single session.
>>>=20
>>> Muthu
>>>=20
>>> |-----Original Message-----
>>> |From: dispatch-bounces@ietf.org=20
>> [mailto:dispatch-bounces@ietf.org] On
>>> Behalf
>>> |Of Cullen Jennings (fluffy)
>>> |Sent: Wednesday, June 09, 2010 8:23 PM
>>> |To: DISPATCH list
>>> |Subject: Re: [dispatch] Final Charter for Session-ID?
>>> |
>>> |I have a problem with this proposed charter.
>>> |
>>> |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
>>> |
>>> |>
>>> |> The definition of a higher-layer "session" of the same=20
>> "message-flow"
>>> is
>>> |not defined by SIP when it involves a B2BUA, and is difficult to
>>> normatively
>>> |define in practice.  For the purposes of this WG charter's=20
>> scope, two
>>> or
>>> |more dialogs of a B2BUA are correlated under any conditions where=20=

>>> |correlation might aid troubleshooting, by identifying them as=20
>>> |belonging
>>> to
>>> |the same higher-level session. Any solution documents from this=20
>>> |working group may expand or constrain the scope of their=20
>> mechanism's
>>> correlation, if
>>> |the working group so decides.
>>> |
>>> |I think the above text is way too vague and impossible to=20
>> reduce to=20
>>> |engineering practice. If you have 10 calls going to a PSTN=20
>> GWs that=20
>>> |is having voice quality problems, clearly having all of=20
>> them grouped
>>> together
>>> |is useful in debugging. Yet I doubt we want all the calls from a=20
>>> |given
>>> PSTN
>>> |GW to have the same correlation ID. Our lack of any solid=20
>> definition=20
>>> |of
>>> what
>>> |it is we are trying to correlate is a strong indication to=20
>> me that we
>>> have
>>> |not yet come to any consensus about what the problem is=20
>> this WG would
>>> be
>>> |trying to fix.
>>> |
>>> |I would like to see a WG charted in this area but I would=20
>> expect the
>>> IESG to
>>> |more or less laugh at a problem statement that reduced down to "uh
>>> might be
>>> |useful for debuggin". Can some people please try and propose a more
>>> concrete
>>> |definition of when two sip dialogs will be correlated?
>>> |
>>> |Cullen
>>> |
>>> |_______________________________________________
>>> |dispatch mailing list
>>> |dispatch@ietf.org
>>> |https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>=20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From fluffy@cisco.com  Tue Jun 15 07:36:43 2010
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To: "Dale R (Dale) WORLEY" <dworley@avaya.com>
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Let Mary and I try and to reserver 30 minutes or more for Alert-Info =
URN. If ALert URN is a WG by that point, it can run it as a very short =
WG meeting and if is not a WG we can use the time to discuss how to get =
to being a WG. We still need to sort out the agenda but I don't foresee =
a problem with this.=20


On Jun 13, 2010, at 11:55 PM, Gonzalo Camarillo wrote:

> Hi,
>=20
> if you need face-to-face time for this, go ahead talk to the DISPATCH
> chairs so that you can either get a slot within the DISPATCH session =
or,
> alternatively, have an ad-hoc meeting. If we managed to charter the WG
> in time, we would have a WG meeting instead.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> On 14/06/2010 5:37 AM, WORLEY, Dale R (Dale) wrote:
>> ________________________________________
>> From: Gonzalo Camarillo [Gonzalo.Camarillo@ericsson.com]
>>=20
>> yes, Mary's description is correct. The process is as follows. Once =
we
>> agreed on the charter, I requested the IESG to review it. This =
charter
>> will be discussed in the next IESG telechat (the coming Thursday),
>> actually. Once the IESG is happy with it, the next step will be to =
have
>> an IETF-wide review. After that, we will be able to charter the WG.
>>=20
>> In any case, as we have said a number of times, proponents of a
>> particular piece of work should not stop working on it while we =
charter
>> their WG. They are encouraged to continue making progress in parallel
>> with the chartering process.
>> ________________________________________
>>=20
>> That is all fine, but what are the implications regarding getting a =
time slot at Maastricht?  There seem to be two choices:  a slot within =
Dispatch, or a separately scheduled Salud session.  What we do not want =
to get stuck with is not being able to get time in Dispatch (because a =
charter has been proposed) and not being able to get a separate session =
(because the charter hasn't gone through all the approval steps), as =
that would prevent any session at  Maastricht.
>>=20
>> Dale
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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Thanks - I changed that a little bit and put it in the updated version.


On Jun 2, 2010, at 12:49 PM, Peter Musgrave wrote:

> How about:
>=20
> "The essential characteristic of ViPR is establishing authentication
> by PSTN reachability and not by e.g. direct reference to ENUM
> databases or other assertions of PSTN number ownership. Elements such
> as public ENUM may be employed indirectly as part of assessing the
> PSTN reachability but no direct interaction with ENUM will be
> required. "
>=20
> Peter
>=20
> On Wed, Jun 2, 2010 at 2:35 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>>=20
>> I kept trying to write something about the relation of ViPR to ENUM =
but it kept coming out sounding like public ENUM had failed - which was =
not really what I wanted to say so I eventually deleted it. If someone =
had some good text about ENUM, would be nice to add it.
>>=20
>> I view infrastructure and carrier ENUM as somewhat unrelated to this =
but public ENUM is related. However, ENUM talks about how to query the =
database, but it is somewhat silent on how data gets in the database. =
How does the database authorize that someone is authoritative for a =
number and can put write a record into the ENUM database. VIPR is more =
about that side of the problem.
>>=20
>> And +1 to Paul's answers of the other questions ....
>>=20
>> On Jun 1, 2010, at 12:28 PM, Peter Musgrave wrote:
>>=20
>>> Hi Cullen,
>>>=20
>>> Does the charter need to say anything explicitly about not relying =
on
>>> any of the work which uses ENUMs in DNS as a way of associating SIP
>>> endpoints and IP addresses?
>>>=20
>>> AFAIK ViPR is about *using* a SIP call to determine a path to the
>>> endpoint (however that happens to work).
>>>=20
>>> Does the charter need to stipulate that the mapping from a PSTN =
number
>>> to a SIP device is a one-to-one mapping? Does there need to be text
>>> about how multiple endpoints which are behind a border device (and =
all
>>> appear as the same PSTN number) are to be handled?.
>>>=20
>>> ViPR so far describes an "edge-device" approach. Is the charter
>>> restricted to this approach?
>>>=20
>>> Thanks,
>>>=20
>>> Peter Musgrave
>>>=20
>>>=20
>>>=20
>>> On Tue, Jun 1, 2010 at 12:55 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>>>>=20
>>>> I've been talking to a lot of people about the VIPR drafts  - here =
is a first cut of a proposal for a WG that could do this. I'm sure the =
charter proposal needs a bunch of work but I wanted to get the =
discussion rolling on the list.
>>>>=20
>>>> Thanks, Cullen
>>>>=20
>>>> (PS - this is sent in my individual contributor role. Take all my =
posts about VIPR to be in my individual role not my co-chair role)
>>>>=20
>>>> -------------------------------------------------------
>>>>=20
>>>> ViPR Charter Proposal (Version 0)
>>>>=20
>>>> WG Name:  Verification Involving PSTN Reachability (VIPR)
>>>>=20
>>>> There are two globally deployed address spaces for communications =
that more than a billion people use on a daily basis. They are phone =
numbers and DNS rooted address such as web servers and email addresses. =
The federation design of SIP is primarily designed for email style =
addresses yet a large percentage of SIP deployments primarily use phone =
numbers for identifying users. The goal of this working group is to =
allows people to use SIP to federate over the the internet while still =
using phone numbers to identify the person they wish to communicate =
with.
>>>>=20
>>>> The VIPR WG will address this problem by developing a peer to peer =
based approach to finding SIP domains that claim to be responsible for a =
given phone number and the WG will design validation protocols to ensure =
a reasonable likelihood that a given domain actually is responsible for =
the phone number. One initial validation protocol will be based on a =
domain being able to prove it received a particular phone call over the =
PSTN based on both sides knowing the start and stop times of that call. =
Other validation schemes, such as examining fingerprints or watermarking =
of PSTN media, to show that a domain received a particular PSTN phone =
call may be considered by the working group. To help mitigate SPAM over =
SIP issues, the WG will define an token based authorization scheme so =
that domain using SIP to federate can choose to check that incoming SIP =
calls are from a domain that successfully validated a phone number.
>>>>=20
>>>> The problem statement and some possible starting points for =
solutions are further desired in the following internet drafts which =
shall form the bases of the WG documents:
>>>> draft-rosenberg-dispatch-vipr-overview
>>>> draft-rosenberg-dispatch-vipr-reload-usage
>>>> draft-rosenberg-dispatch-vipr-pvp
>>>> draft-rosenberg-dispatch-vipr-sip-antispam
>>>>=20
>>>> The working group will carefully coordinate with the security area, =
O&M area, as well as the appropriate RAI WG including sipcore and =
p2psip.
>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>=20
>>>=20
>>=20
>>=20
>> Cullen Jennings
>> For corporate legal information go to:
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>=20
>>=20
>>=20
>>=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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On Jun 2, 2010, at 12:31 PM, Marc Petit-Huguenin wrote:

>>=20
>> The problem statement and some possible starting points for solutions =
are further desired in the following internet drafts which shall form =
the bases of the WG documents:
>> draft-rosenberg-dispatch-vipr-overview
>> draft-rosenberg-dispatch-vipr-reload-usage
>> draft-rosenberg-dispatch-vipr-pvp
>> draft-rosenberg-dispatch-vipr-sip-antispam
>=20
> VAP is not listed here.  Is it an oversight or is it because the WG =
will work
> only on interoperability between ViPR servers?

I could go either way on VAP. On one hand, the minimal thing we need to =
standardize to have interoperability between domain is just the above =
stuff without VAP. On the other hand, VAP is a very simple way to =
modularize what is happening inside a domain. VAP allows information =
about PSTN call to be given to the server doing the VIPR stuff and =
allows the VIPR server to tell routing engines and PBX inside the domain =
about new routes.=20

I'd be interested in hearing what other thing. It's easy to add =
something like VAP to the charter.=20


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From: Cullen Jennings <fluffy@cisco.com>
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To: Jean-Francois Mule <jf.mule@cablelabs.com>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN Reachability (VIPR)
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Hi - many thanks for the great comments - few comments inline but I =
agree with all your points and tried to update the charter =
appropriately.=20

On Jun 2, 2010, at 2:59 PM, Jean-Francois Mule wrote:

> Hi,
>=20
>   I support this proposed work to be taken by IETF under a new working =
group.  There are certainly concrete deliverables that would merit IETF =
consensus to spur multi-vendor interoperability.
>=20
>  A few comments below, mostly to help bash the proposed charter.
>=20
> Jean-Francois.
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
>> On Behalf Of Cullen Jennings
>> Sent: Tuesday, June 01, 2010 10:55 AM
>> To: DISPATCH list
>> Subject: [dispatch] Charter Proposal: Verification Involving PSTN
>> Reachability (VIPR)
>>=20
>>=20
>> I've been talking to a lot of people about the VIPR drafts  - here
>> is a first cut of a proposal for a WG that could do this. I'm sure
>> the charter proposal needs a bunch of work but I wanted to get the
>> discussion rolling on the list.
>>=20
>> Thanks, Cullen
>>=20
>> (PS - this is sent in my individual contributor role. Take all my
>> posts about VIPR to be in my individual role not my co-chair role)
>>=20
>> -------------------------------------------------------
>>=20
>> ViPR Charter Proposal (Version 0)
>>=20
>> WG Name:  Verification Involving PSTN Reachability (VIPR)
>>=20
>> There are two globally deployed address spaces for communications
>> that more than a billion people use on a daily basis. They are
>> phone numbers and DNS rooted address such as web servers and email
>> addresses. The federation design of SIP is primarily designed for
>                 ^^^^^^^^^^^^^^^^^^^^^^^^
> Given some folks talk about SIP federations in the context of =
speermint (http://tools.ietf.org/html/rfc5486#section-5) and I don't =
think this is what you mean here, this choice of terms may not be the =
best.  As much as some folks disagree on the worthiness of the output of =
speermint, at least SIP federations are described there.  Would =
recommend clarifying this somehow.

yah - agree. "Federation" means a lot of things but conflicting with =
speermint terminology seems like a bad plan. I'll try and rephrase in =
terms of "inter domain calling"=20

>=20
>> email style addresses yet a large percentage of SIP deployments
>> primarily use phone numbers for identifying users. The goal of this
>> working group is to allows people to use SIP to federate over the
>                                       ^^^^^^^^^^^^^^^^^^^
> my understanding is that those people would not only use SIP but some =
other smarts to put some stuff in DHTs to enable PSTN reachability =
verification. This implies that the solution solely relies on SIP.  May =
want to expand or clarify, or else qualify the sentence further.

yep ... I think this works for more than just SIP

>=20
>> the internet while still using phone numbers to identify the person
>> they wish to communicate with.
>>=20
>=20
>=20
>> The VIPR WG will address this problem by developing a peer to peer
>> based approach to finding SIP domains that claim to be responsible
>                            ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> That may open a rathole: what does it mean to be responsible for a =
given phone number (worst case, Rich Shockey will chime in and will =
muddy this more, or John Elwell will claim this should have been in =
speermint but it cannot be found there...).
>=20
> Do you mean any SIP Service Provider =
(http://tools.ietf.org/html/rfc5486#section-3.9) on the signaling path =
of a SIP invite or dialog-initiating request for that telephone number?  =
Do you mean the SIP domain responsible for the last hop before the SIP =
UA? Both?  Or is it the latter with the pre-requisite that PSTN =
connectivity is required somehow?

It's was deliberately a bit vague. The real goal is to get the best =
approximation we can of the administrative domain who controls the phone =
that rings when you dial that number. The whole concept of ownership of =
a phone number is so muddled that trying to use owner is clearly =
useless. It's very unclear if my cell phone number is owned by me (I can =
LNP to wherever I want and Cisco lets me take on termination of my =
employment), to Cisco (they pay the bill), to AT&T, to Neustar, or to =
some government, or any of the other parties that could cause it not to =
work. I think the best definition we can come up with of responsible is =
an administrative domain that is always involved in the call signaling =
of a PSTN call to that phone number. I tired to clear up in the charter =
a bit but this is going to need a bunch of people to help fine tune =
text.=20

>=20
>> for a given phone number and the WG will design validation
>> protocols to ensure a reasonable likelihood that a given domain
>> actually is responsible for the phone number. One initial
>> validation protocol will be based on a domain being able to prove
>> it received a particular phone call over the PSTN based on both
>> sides knowing the start and stop times of that call. Other
> This last sentence is advocating one specific solution. =20
> I think the VIPR drafts are powerful and this may be part of a working =
solution.  But at this stage in the WG charter definition, would it be =
preferable to say something like:
> 	Some validation protocols may be based on additional knowledge =
gathered around a SIP call, for example, the ability to prove a call was =
received over the PSTN based on start and stop times.
>=20
> It leave things open and welcomes other proposals.  It also does not =
preclude the validation protocol you have documented.

Fair enough - I changed draft charter to you suggested text.  If we =
think there is a reasonable chance that we might not do this start/stop =
time validations, then this change makes sense but if we are pretty =
certain we will do this approach, then I generally prefer making the =
charters as specific as possible.=20

>=20
>> validation schemes, such as examining fingerprints or watermarking
>> of PSTN media, to show that a domain received a particular PSTN
>> phone call may be considered by the working group.=20
> These are additional examples that must be part of the same bucket as =
the one before imo.  Currently it reads that PSTN+start+stop is the =
initial one, others may be considered later. =20
> I have no preference and like the elegance of the approach in some the =
VIPR drafts.  But the charter should make clear that the WG will decide =
this.

OK

>=20
>> To help mitigate
>> SPAM over SIP issues, the WG will define an token based
>> authorization scheme so that domain using SIP to federate can
>> choose to check that incoming SIP calls are from a domain that
>> successfully validated a phone number.
> Same here, you state a solution to a pb.  I would prefer to have a =
requirement in the charter that mandates the WG comes up with a method =
to help mitigate SPIT by ensuring that a domain using SIP can validate =
incoming calls are indeed from a domain that successfully validated the =
TN.

Ok - changed to say that.=20

>=20
>=20
>> The problem statement and some possible starting points for
>> solutions are further desired in the following internet drafts
>> which shall form the bases of the WG documents:
>> draft-rosenberg-dispatch-vipr-overview
>> draft-rosenberg-dispatch-vipr-reload-usage
>> draft-rosenberg-dispatch-vipr-pvp
>> draft-rosenberg-dispatch-vipr-sip-antispam
>>=20
>> The working group will carefully coordinate with the security area,
>> O&M area, as well as the appropriate RAI WG including sipcore and
>> p2psip.
>>=20
>>=20
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch



From fluffy@cisco.com  Tue Jun 15 10:24:49 2010
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Subject: [dispatch] VIPR - proposed charter version 2
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Thanks for all the comments on the list. Here is a revised version:

(PS. If anyone else wants to edit, this is in SVN and glad to get anyone =
write permission)

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

ViPR Charter Proposal (Version 2)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications that
more than a billion people use on a daily basis. They are phone numbers
and DNS rooted address such as web servers and email addresses. The
inter-domain signaling design of SIP is primarily designed for email
style addresses yet a large percentage of SIP deployments mostly use
phone numbers for identifying users. The goal of this working group is
to enable inter-domains communications over the the internet, using
protocol such as SIP, while still allowing people to use phone numbers
to identify the person they wish to communicate with.

The VIPR WG will address this problem by developing a peer to peer based
approach to finding domains that claim to be responsible for a given
phone number and validation protocols to ensure a reasonable likelihood
that a given domain actually is responsible for the phone number. In
this context, we use "responsible" to mean an administrative domain
would would be at least one of the administrative domains that a PSTN
call to this phone number would be routed to. Once the domain
responsible for the phone number is found, existing protocols such as
SIP and XMPP can be used for inter-domain communications.

Some validation protocols may be based on knowledge gathered around a
SIP call, for example, the ability to prove a call was received over the
PSTN based on start and stop times. Other validation schemes, such as
examining fingerprints or watermarking of PSTN media, to show that a
domain received a particular PSTN phone call may also be considered by
the working group. The WG will select and standardize at least one
validation scheme.

To help mitigate SPAM issues when using SIP between domains, the WG will
define a mechanisms to enable one domain to check that incoming SIP
messages from a different domain are coming from a domain that
successfully validated a phone number. The working group will define new
SIP headers and option tags to enable this.

The essential characteristic of ViPR is establishing authentication by
PSTN reachability when it is not possible to use a direct reference to
ENUM databases or other direct assertions of PSTN number
ownership. Elements such as public ENUM easily coexist with VIPR but no
direct interaction with ENUM will be required.

The problem statement and some possible starting points for solutions
are further discussed in the following internet drafts which shall form
the bases of the WG documents:

draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam

The working group will carefully coordinate with the security area, O&M
area, as well as the appropriate RAI WG including sipcore and p2psip.




From dworley@avaya.com  Tue Jun 15 10:28:10 2010
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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 15 Jun 2010 13:27:59 -0400
Thread-Topic: Conceptual proposal for extensibility of Alert-Info URNs
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Subject: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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[As I am the chair-elect for this topic, I now have to identify that I
am proposing this as an individual contributor-elect.]

This is a conceptual proposal for for specifying Alert-Info URNs.  The
problem that I am attacking is extensibility -- because the practical
application of the Alert-Info URNs will be as successors of the
ring tone and ringback tone implementations of innumerable existing
telephone systems, we expect the uses of Alert-Info URNs to be subject
to much extension.  The primary problem is to provide a framework
which allows such extensions to be implemented without damaging
interoperability between systems whose Alert-Info usage has not been
coordinated a priori.  A secondary problem is to ensure that the
framework can be implemented efficiently in practical situations.

The basis of this proposal has been stolen from previous contributors
to the Alert-Info work.

Since this is a conceptual proposal, I am not particular about the
syntax to be used -- the syntax shown here is just to illustrate the
concepts.  What I am proposing is a system of semantics; many variant
syntaxes can be used to implement it with differing tradeoffs.

* Terminology

We say that a "specifier" sends an "indication" (a URN in an
Alert-Info header) to a "renderer" which then "renders" a "signal"
based on the indication to a human user.  A "category" is a
characteristic whose "values" can be used to classify indications.

* Requirements and non-requirements

Requirement A:  Indications must be able to represent a wide variety
of signals, which have many largely-orthogonal characteristics.  (See
draft-liess-dispatch-alert-info-urns-00 section section 2 for a
partial list, and Paul Kyzivat's message for another partial list.)

Requirement B:  Indications include subsets which have distinctly
different semantics, that is, they form disjoint "value spaces".  For
example, some indications should describe the semantics of the
signaling situation whereas others should describe the audio
characteristics of the signal.  This implies that there is no single
set of categories that can be used as independent coordinates of the
value-space of indications.

Requirement C:  The set of indications must be able to support
extensibility by a wide variety of organizations that are not
coordinated with each other.  Extensions must be able to:

- add further values to any existing category

- add further categories that are orthogonal to existing categories

- semantically subdivide the meaning provided by any existing
  indication

- add further "value spaces" of indications whose semantics are not
  related to existing indications, and thus, whose categories differ
  from and do not interact with existing categories

Requirement D:  An indication is transmitted from a specifier to a
renderer, which must base its rendering decision only on the
indication.  In particular, there is no multi-message negotiation
process or carrying of context from one indication to the next.

Requirement E:  The renderer may be customized in ways that limit the
set of signals that it can render, or it may be provided with a set of
signals that have uncommon semantics.  (The canonical example is a UA
for the hearing-impaired.)  (By Requirement D, the renderer has no way
of transmitting this fact to the specifier.)

Requirement F:  If the specifier and the renderer have designs that
are properly coordinated, the indications must be able to reliably
carry all extensions that are supported in the coordinated designs.

Requirement G:  In any situation, the renderer must be able to perform
close to the best possible rendering that it could do even the
specifier had specific knowledge of the renderer's capabilities.

Non-Requirement H:  In any situation other than that of Requirement F,
we do not require the render to perform the best possible rendering.

* Categories, values, and indicators

There are a theoretically infinite number of categories.  Each
category is named by a word, such as:

    reason
    priority
    source
    duration

The values of each category conceptually form a tree.  The root of the
tree is the universal, or most general, value.  It is denoted by the
the category name, and represents the universe of all possibilities.
Each further subdivision of a value is specified by adding a hyphen
and another word.  Thus, within the category "state" we have values:

    state
    state-unknown
    state-waiting
    state-on.call
    state-away

(Here we pretend '.' is a letter.)

The first of these values specifies no specific information.  The
following values specify subdivisions of the semantic space.  The
value "state-waiting" can be further subdivided based on the
importance of the call being waited upon:

    state-waiting-standard
    state-waiting-vip

An indication is assembled by providing a sequence of values (which
implicitly specify their categories), separated by colons, in order of
importance.  That is, the first value is the one that should be
rendered as accurately as possible; within that constraint, the second
value should be rendered as accurately as possible; etc.

For example, we can assemble the following indications:

    an internal, priority call
        source-internal:priority-high

    priority call waiting
        reason-call.waiting:priority-high

    short Albanian auto-callback
        reason-auto.callback:duration-short:locale-al

(The specific examples are taken from
draft-liess-dispatch-alert-info-urns-00 section 4.5, the specific
categories and values are adapted from Paul Kyzivat's message.)

All of these indications are easily extensible.  For instance, if
Albania has two service providers with different signal tones, we
could specialize the last indication into:

    short Albanian auto-callback, service provider A
        reason-auto.callback:duration-short:locale-al-sp.A

    short Albanian auto-callback, service provider B
        reason-auto.callback:duration-short:locale-al-sp.B

Due to the rule that the earlier values in an indicator have more
influence than latter values, there is a subtle difference between:

    source-internal:priority-high
    priority-high:source-internal

in regard to which aspect of the indication it is more important to
render.

* Selecting the signal based on an indication

Having defined a universe of indications with infinite extensibility,
we now need to show that a practical device can interpret indications
to select from a limited set of signals.  To do this, I will describe
a very general algorithm for doing this selection in all possible
situations.  Note that in practical cases (where all the signals that
can be rendered can be described by a small number of values of a
small number of categories) the algorithm will operate in a reasonably
rapid and simple way.

To start with, we assume that the renderer possesses a set of
available signals.  This set will naturally be partially determined by
the renderer itself, but the set may also be affected by the
configuration of the renderer (e.g., hearing-impaired mode vs. hearing
mode) and the context of the indication (e.g., ring tone vs. ringback
tone situations).  The set may be finite or infinite.  Of course, if
the set is infinite, it can't be represented literally, but the
renderer necessarily has a method of representing the set and its
possible subsets.  A finite set of available signals can be enumerated
directly, or represented in some other way.

Each available signal is described by the values of a set of
categories.  We assume that the renderer only knows the values for a
finite set of categories; for all other categories, all available
signals implicitly have the "universal", root value.

Informally, the algorithm proceeds as follows:

    Load a set of "acceptable signals" with all available signals.

    Process the words in the values in the indication from left to right.
    As each additional word is considered, consider the portion of the
    value that has been examined, and remove from the acceptable set all
    signals which are inconsistent with that value, in that their value
    for that category is not equal to or below the indication's value (as
    processed so far).

        If after processing a word, one or more signals remain in the
        acceptable set, continue processing the next word.

        If after processing a word, no signals remain in the
        acceptable set, restore the previous state of the acceptable
        set and continue with the next value in the indication.

    At the end of processing, the acceptable set will contain one or more
    signals.  Choose from among them according to some sort of priority
    system.

More formally, an indication is processed by:

      initialize the acceptable set of signals with all available signals

      for each value in the indication (going left to right):

          for each word in the value (going left to right, which is down
          the tree of values):

              if there are acceptable signals which have a value of the
              category which is equal to or below the indication value as
              seen so far:

                  delete all acceptable signals which do not have such a
                  value, and continue processing this value

              else

                  cease processing this value and continue with the next
                  value

      the acceptable set will contain at least one signal.  Choose from
      among the acceptable signals via some priority scheme.

Let me work through an example.

Let us suppose that the renderer implements three categories:

    reason
        reason-normal
        reason-call.waiting
        reason-forward
        reason-auto.callback
        reason-recall.hold
        reason-recall.transfer

    duration
        duration-normal
        duration-short
        duration-long

    locale
        locale-fm
        locale-al
            locale-al-sp.A
            locale-al-sp.B

Let us suppose that the renderer has signals for all combinations of
the leaf values of these categories, except that for historical
reasons if "reason" has a non-universal value, "default" must be
universal, and vice-versa; the signals (copied from the PSTN service
providers) do not allow signaling a non-standard reason with a
non-standard duration.  So the allowed combinations of these two
values are:

    reason-normal & duration
    reason-call.waiting & duration
    reason-forward & duration
    reason-auto.callback & duration
    reason-recall.hold & duration
    reason-recall.transfer & duration
    reason & duration-normal
    reason & duration-short
    reason & duration-long

There are signals that combine each of these with "locale-fm",
"locale-al-sp.A", and "locale-al-sp.B".

To process the indication
"reason-auto.callback:duration-short:locale-al-sp.A", the renderer
first loads the acceptable signal set with all 27 of its signals, and
then examines the indication word by word:

- "reason"

Removes no signals from the set.

- "reason-auto.callback"

Reduces the set to

    reason-auto.callback & duration & locale-fm
    reason-auto.callback & duration & locale-al-sp.A
    reason-auto.callback & duration & locale-al-sp.B

- "duration"

Removes no signals from the set.

- "duration-short"

Since no signals in the set have a value at or under this value, no
change is made and the processing of this value ends.

- "locale"

Removes no signals from the set.

- "locale-al"

Reduces the set to

    reason-auto.callback & duration & locale-al-sp.A
    reason-auto.callback & duration & locale-al-sp.B

- "locale-al-sp.A"

Reduces the set to

    reason-auto.callback & duration & locale-al-sp.A

The end of the indication is reached, and since only one signal remains
in the set, it is rendered.

Now, if the indication was less specific,
"reason-auto.callback:duration-short:locale-al", at the end the
acceptable set would be

    reason-auto.callback & duration & locale-al-sp.A
    reason-auto.callback & duration & locale-al-sp.B

and the renderer would have to choose between these two signals in
some way.

Let us suppose that the renderer implemented a further category:

    mode
        mode-visual
        mode-audio

and applied the value "mode-audio" to all of the above signals, but
only added the following signals with "mode-visual":

    reason & duration-normal & locale & mode-visual
    reason & duration-short & locale & mode-visual
    reason & duration-long & locale & mode-visual

That is, the only variation provided is in "duration".

Then the indication "reason-auto.callback:duration-short:locale-al"
would generate the same acceptable set as above, that is, there would
be no "mode-visual" signals.  The indication
"mode-visual:reason-auto.callback:duration-short:locale-al" would
generate the acceptable set:

    reason & duration-short & locale & mode-visual

This is because the indication implies that matching "mode" is more
important than matching "reason" and "duration".  Once the selection
for "mode-visual" has been done, no further selection can be done on
"reason", but "duration-short" can be acted upon.

In practice, for use with a hearing-impaired user, the initial set of
signals would be restricted to those with "mode-visual".  In this
example, we have assumed that the user considers "mode-visual" and
"mode-audio" to be equally acceptable.

If we configured the renderer to only use "mode-visual" signals, then
the indication "reason-auto.callback:duration-short:locale-al" would
select only the signal

    reason & duration-short & locale & mode-visual

In this configuration, the indication
"mode-audio:reason-auto.callback:duration-short:locale-al" would also
select that signal, because the value "mode-audio" cannot be acted
upon within that set of available signals.  It is effectively ignored.

* Extensibility

Let us now see how this scheme supports the various types of
extensibility listed in Requirement C.

- add further values to any existing category

We could define an additional value such as

    reason-avaya.special

If the renderer had any signals that had this value, they would be
selected by an indicator containing "reason-avaya.special".  If the
renderer has no such signals, then in an indicator this value acts the
same as "reason", that is, it has no effect.

- add further categories that are orthogonal to existing categories

We could define an additional category with values:

    avaya
        avaya-a
        avaya-b
        avaya-c

If a renderer does not know of this category, all of its signals
implicitly have the value "avaya".  Because of this, when such a
renderer processes a value of this category in an indicator, it never
changes the available set.

- semantically subdivide the meaning provided by any existing
  indication

We could define subvalues of existing values, such as:

    state
        state-waiting
            state-waiting-avaya.standard
            state-waiting-avaya.vip

If the renderer has two signals that differ only in the values
"state-waiting-avaya.standard" and "state-waiting-avaya.vip", any
indicator that does not know of this subdivision will never separate
the two signals; they will both remain in the acceptable set or both
be excluded from it.  So the renderer needs a policy for choosing one
over the other -- In this case, clearly "state-waiting-avaya.standard"
is to be preferred.

However, if the indicator gives one of the two subvalues, the renderer
will select the appropriate signal.

If the indicator gives one of the two subvalues but the renderer only
knows of "state-waiting", the "avaya.standard" or "avaya.vip" part of
the value will never affect the acceptable set (since to select based
on it would make the acceptable set empty).  Processing of
"state-waiting-avaya.standard" would have the same effect as
"state-waiting".

- add further "value spaces" of indications whose semantics are not
  related to existing indications, and thus, whose categories differ
  from and do not interact with existing categories

We can define completely independent spaces of signals by defining new
categories to describe the new space, giving signals in one space the
universal value for categories describing the other space.

For example, we can define

    auto
        auto-make
            auto-make-volvo
            auto-make-toyota
    color
        color-red
        color-green

If we have signals with the values

    auto-make-volvo & color-red
    auto-make-volvo & color-green
    auto-make-toyota & color-red
    auto-make-toyota & color-green

and add them to the set of "mode-visual" signals given above, we get:

    reason & duration-normal & locale & mode-visual & auto & color
    reason & duration-short & locale & mode-visual & auto & color
    reason & duration-long & locale & mode-visual & auto & color
    reason & duration & locale & mode & auto-make-volvo & color-red
    reason & duration & locale & mode & auto-make-volvo & color-green
    reason & duration & locale & mode & auto-make-toyota & color-red
    reason & duration & locale & mode & auto-make-toyota & color-green

This set is a combination of chalk and cheese.  But the first
non-universal value in the indicator will select signals of one type
or the other.  For instance, "auto-make" will select

    reason & duration & locale & mode & auto-make-volvo & color-red
    reason & duration & locale & mode & auto-make-volvo & color-green
    reason & duration & locale & mode & auto-make-toyota & color-red
    reason & duration & locale & mode & auto-make-toyota & color-green

and "duration-normal" will select

    reason & duration-normal & locale & mode-visual & auto & color

Dale

From pkyzivat@cisco.com  Tue Jun 15 11:20:20 2010
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Very interesting!

Without getting into syntax specifics, I think I like this - it provides 
a much cleaner conceptual framework. The idea of the specifier providing 
a priority seems to resolve some loose ends.

	Thanks,
	Paul

WORLEY, Dale R (Dale) wrote:
> [As I am the chair-elect for this topic, I now have to identify that I
> am proposing this as an individual contributor-elect.]
> 
> This is a conceptual proposal for for specifying Alert-Info URNs.  The
> problem that I am attacking is extensibility -- because the practical
> application of the Alert-Info URNs will be as successors of the
> ring tone and ringback tone implementations of innumerable existing
> telephone systems, we expect the uses of Alert-Info URNs to be subject
> to much extension.  The primary problem is to provide a framework
> which allows such extensions to be implemented without damaging
> interoperability between systems whose Alert-Info usage has not been
> coordinated a priori.  A secondary problem is to ensure that the
> framework can be implemented efficiently in practical situations.
> 
> The basis of this proposal has been stolen from previous contributors
> to the Alert-Info work.
> 
> Since this is a conceptual proposal, I am not particular about the
> syntax to be used -- the syntax shown here is just to illustrate the
> concepts.  What I am proposing is a system of semantics; many variant
> syntaxes can be used to implement it with differing tradeoffs.
> 
> * Terminology
> 
> We say that a "specifier" sends an "indication" (a URN in an
> Alert-Info header) to a "renderer" which then "renders" a "signal"
> based on the indication to a human user.  A "category" is a
> characteristic whose "values" can be used to classify indications.
> 
> * Requirements and non-requirements
> 
> Requirement A:  Indications must be able to represent a wide variety
> of signals, which have many largely-orthogonal characteristics.  (See
> draft-liess-dispatch-alert-info-urns-00 section section 2 for a
> partial list, and Paul Kyzivat's message for another partial list.)
> 
> Requirement B:  Indications include subsets which have distinctly
> different semantics, that is, they form disjoint "value spaces".  For
> example, some indications should describe the semantics of the
> signaling situation whereas others should describe the audio
> characteristics of the signal.  This implies that there is no single
> set of categories that can be used as independent coordinates of the
> value-space of indications.
> 
> Requirement C:  The set of indications must be able to support
> extensibility by a wide variety of organizations that are not
> coordinated with each other.  Extensions must be able to:
> 
> - add further values to any existing category
> 
> - add further categories that are orthogonal to existing categories
> 
> - semantically subdivide the meaning provided by any existing
>   indication
> 
> - add further "value spaces" of indications whose semantics are not
>   related to existing indications, and thus, whose categories differ
>   from and do not interact with existing categories
> 
> Requirement D:  An indication is transmitted from a specifier to a
> renderer, which must base its rendering decision only on the
> indication.  In particular, there is no multi-message negotiation
> process or carrying of context from one indication to the next.
> 
> Requirement E:  The renderer may be customized in ways that limit the
> set of signals that it can render, or it may be provided with a set of
> signals that have uncommon semantics.  (The canonical example is a UA
> for the hearing-impaired.)  (By Requirement D, the renderer has no way
> of transmitting this fact to the specifier.)
> 
> Requirement F:  If the specifier and the renderer have designs that
> are properly coordinated, the indications must be able to reliably
> carry all extensions that are supported in the coordinated designs.
> 
> Requirement G:  In any situation, the renderer must be able to perform
> close to the best possible rendering that it could do even the
> specifier had specific knowledge of the renderer's capabilities.
> 
> Non-Requirement H:  In any situation other than that of Requirement F,
> we do not require the render to perform the best possible rendering.
> 
> * Categories, values, and indicators
> 
> There are a theoretically infinite number of categories.  Each
> category is named by a word, such as:
> 
>     reason
>     priority
>     source
>     duration
> 
> The values of each category conceptually form a tree.  The root of the
> tree is the universal, or most general, value.  It is denoted by the
> the category name, and represents the universe of all possibilities.
> Each further subdivision of a value is specified by adding a hyphen
> and another word.  Thus, within the category "state" we have values:
> 
>     state
>     state-unknown
>     state-waiting
>     state-on.call
>     state-away
> 
> (Here we pretend '.' is a letter.)
> 
> The first of these values specifies no specific information.  The
> following values specify subdivisions of the semantic space.  The
> value "state-waiting" can be further subdivided based on the
> importance of the call being waited upon:
> 
>     state-waiting-standard
>     state-waiting-vip
> 
> An indication is assembled by providing a sequence of values (which
> implicitly specify their categories), separated by colons, in order of
> importance.  That is, the first value is the one that should be
> rendered as accurately as possible; within that constraint, the second
> value should be rendered as accurately as possible; etc.
> 
> For example, we can assemble the following indications:
> 
>     an internal, priority call
>         source-internal:priority-high
> 
>     priority call waiting
>         reason-call.waiting:priority-high
> 
>     short Albanian auto-callback
>         reason-auto.callback:duration-short:locale-al
> 
> (The specific examples are taken from
> draft-liess-dispatch-alert-info-urns-00 section 4.5, the specific
> categories and values are adapted from Paul Kyzivat's message.)
> 
> All of these indications are easily extensible.  For instance, if
> Albania has two service providers with different signal tones, we
> could specialize the last indication into:
> 
>     short Albanian auto-callback, service provider A
>         reason-auto.callback:duration-short:locale-al-sp.A
> 
>     short Albanian auto-callback, service provider B
>         reason-auto.callback:duration-short:locale-al-sp.B
> 
> Due to the rule that the earlier values in an indicator have more
> influence than latter values, there is a subtle difference between:
> 
>     source-internal:priority-high
>     priority-high:source-internal
> 
> in regard to which aspect of the indication it is more important to
> render.
> 
> * Selecting the signal based on an indication
> 
> Having defined a universe of indications with infinite extensibility,
> we now need to show that a practical device can interpret indications
> to select from a limited set of signals.  To do this, I will describe
> a very general algorithm for doing this selection in all possible
> situations.  Note that in practical cases (where all the signals that
> can be rendered can be described by a small number of values of a
> small number of categories) the algorithm will operate in a reasonably
> rapid and simple way.
> 
> To start with, we assume that the renderer possesses a set of
> available signals.  This set will naturally be partially determined by
> the renderer itself, but the set may also be affected by the
> configuration of the renderer (e.g., hearing-impaired mode vs. hearing
> mode) and the context of the indication (e.g., ring tone vs. ringback
> tone situations).  The set may be finite or infinite.  Of course, if
> the set is infinite, it can't be represented literally, but the
> renderer necessarily has a method of representing the set and its
> possible subsets.  A finite set of available signals can be enumerated
> directly, or represented in some other way.
> 
> Each available signal is described by the values of a set of
> categories.  We assume that the renderer only knows the values for a
> finite set of categories; for all other categories, all available
> signals implicitly have the "universal", root value.
> 
> Informally, the algorithm proceeds as follows:
> 
>     Load a set of "acceptable signals" with all available signals.
> 
>     Process the words in the values in the indication from left to right.
>     As each additional word is considered, consider the portion of the
>     value that has been examined, and remove from the acceptable set all
>     signals which are inconsistent with that value, in that their value
>     for that category is not equal to or below the indication's value (as
>     processed so far).
> 
>         If after processing a word, one or more signals remain in the
>         acceptable set, continue processing the next word.
> 
>         If after processing a word, no signals remain in the
>         acceptable set, restore the previous state of the acceptable
>         set and continue with the next value in the indication.
> 
>     At the end of processing, the acceptable set will contain one or more
>     signals.  Choose from among them according to some sort of priority
>     system.
> 
> More formally, an indication is processed by:
> 
>       initialize the acceptable set of signals with all available signals
> 
>       for each value in the indication (going left to right):
> 
>           for each word in the value (going left to right, which is down
>           the tree of values):
> 
>               if there are acceptable signals which have a value of the
>               category which is equal to or below the indication value as
>               seen so far:
> 
>                   delete all acceptable signals which do not have such a
>                   value, and continue processing this value
> 
>               else
> 
>                   cease processing this value and continue with the next
>                   value
> 
>       the acceptable set will contain at least one signal.  Choose from
>       among the acceptable signals via some priority scheme.
> 
> Let me work through an example.
> 
> Let us suppose that the renderer implements three categories:
> 
>     reason
>         reason-normal
>         reason-call.waiting
>         reason-forward
>         reason-auto.callback
>         reason-recall.hold
>         reason-recall.transfer
> 
>     duration
>         duration-normal
>         duration-short
>         duration-long
> 
>     locale
>         locale-fm
>         locale-al
>             locale-al-sp.A
>             locale-al-sp.B
> 
> Let us suppose that the renderer has signals for all combinations of
> the leaf values of these categories, except that for historical
> reasons if "reason" has a non-universal value, "default" must be
> universal, and vice-versa; the signals (copied from the PSTN service
> providers) do not allow signaling a non-standard reason with a
> non-standard duration.  So the allowed combinations of these two
> values are:
> 
>     reason-normal & duration
>     reason-call.waiting & duration
>     reason-forward & duration
>     reason-auto.callback & duration
>     reason-recall.hold & duration
>     reason-recall.transfer & duration
>     reason & duration-normal
>     reason & duration-short
>     reason & duration-long
> 
> There are signals that combine each of these with "locale-fm",
> "locale-al-sp.A", and "locale-al-sp.B".
> 
> To process the indication
> "reason-auto.callback:duration-short:locale-al-sp.A", the renderer
> first loads the acceptable signal set with all 27 of its signals, and
> then examines the indication word by word:
> 
> - "reason"
> 
> Removes no signals from the set.
> 
> - "reason-auto.callback"
> 
> Reduces the set to
> 
>     reason-auto.callback & duration & locale-fm
>     reason-auto.callback & duration & locale-al-sp.A
>     reason-auto.callback & duration & locale-al-sp.B
> 
> - "duration"
> 
> Removes no signals from the set.
> 
> - "duration-short"
> 
> Since no signals in the set have a value at or under this value, no
> change is made and the processing of this value ends.
> 
> - "locale"
> 
> Removes no signals from the set.
> 
> - "locale-al"
> 
> Reduces the set to
> 
>     reason-auto.callback & duration & locale-al-sp.A
>     reason-auto.callback & duration & locale-al-sp.B
> 
> - "locale-al-sp.A"
> 
> Reduces the set to
> 
>     reason-auto.callback & duration & locale-al-sp.A
> 
> The end of the indication is reached, and since only one signal remains
> in the set, it is rendered.
> 
> Now, if the indication was less specific,
> "reason-auto.callback:duration-short:locale-al", at the end the
> acceptable set would be
> 
>     reason-auto.callback & duration & locale-al-sp.A
>     reason-auto.callback & duration & locale-al-sp.B
> 
> and the renderer would have to choose between these two signals in
> some way.
> 
> Let us suppose that the renderer implemented a further category:
> 
>     mode
>         mode-visual
>         mode-audio
> 
> and applied the value "mode-audio" to all of the above signals, but
> only added the following signals with "mode-visual":
> 
>     reason & duration-normal & locale & mode-visual
>     reason & duration-short & locale & mode-visual
>     reason & duration-long & locale & mode-visual
> 
> That is, the only variation provided is in "duration".
> 
> Then the indication "reason-auto.callback:duration-short:locale-al"
> would generate the same acceptable set as above, that is, there would
> be no "mode-visual" signals.  The indication
> "mode-visual:reason-auto.callback:duration-short:locale-al" would
> generate the acceptable set:
> 
>     reason & duration-short & locale & mode-visual
> 
> This is because the indication implies that matching "mode" is more
> important than matching "reason" and "duration".  Once the selection
> for "mode-visual" has been done, no further selection can be done on
> "reason", but "duration-short" can be acted upon.
> 
> In practice, for use with a hearing-impaired user, the initial set of
> signals would be restricted to those with "mode-visual".  In this
> example, we have assumed that the user considers "mode-visual" and
> "mode-audio" to be equally acceptable.
> 
> If we configured the renderer to only use "mode-visual" signals, then
> the indication "reason-auto.callback:duration-short:locale-al" would
> select only the signal
> 
>     reason & duration-short & locale & mode-visual
> 
> In this configuration, the indication
> "mode-audio:reason-auto.callback:duration-short:locale-al" would also
> select that signal, because the value "mode-audio" cannot be acted
> upon within that set of available signals.  It is effectively ignored.
> 
> * Extensibility
> 
> Let us now see how this scheme supports the various types of
> extensibility listed in Requirement C.
> 
> - add further values to any existing category
> 
> We could define an additional value such as
> 
>     reason-avaya.special
> 
> If the renderer had any signals that had this value, they would be
> selected by an indicator containing "reason-avaya.special".  If the
> renderer has no such signals, then in an indicator this value acts the
> same as "reason", that is, it has no effect.
> 
> - add further categories that are orthogonal to existing categories
> 
> We could define an additional category with values:
> 
>     avaya
>         avaya-a
>         avaya-b
>         avaya-c
> 
> If a renderer does not know of this category, all of its signals
> implicitly have the value "avaya".  Because of this, when such a
> renderer processes a value of this category in an indicator, it never
> changes the available set.
> 
> - semantically subdivide the meaning provided by any existing
>   indication
> 
> We could define subvalues of existing values, such as:
> 
>     state
>         state-waiting
>             state-waiting-avaya.standard
>             state-waiting-avaya.vip
> 
> If the renderer has two signals that differ only in the values
> "state-waiting-avaya.standard" and "state-waiting-avaya.vip", any
> indicator that does not know of this subdivision will never separate
> the two signals; they will both remain in the acceptable set or both
> be excluded from it.  So the renderer needs a policy for choosing one
> over the other -- In this case, clearly "state-waiting-avaya.standard"
> is to be preferred.
> 
> However, if the indicator gives one of the two subvalues, the renderer
> will select the appropriate signal.
> 
> If the indicator gives one of the two subvalues but the renderer only
> knows of "state-waiting", the "avaya.standard" or "avaya.vip" part of
> the value will never affect the acceptable set (since to select based
> on it would make the acceptable set empty).  Processing of
> "state-waiting-avaya.standard" would have the same effect as
> "state-waiting".
> 
> - add further "value spaces" of indications whose semantics are not
>   related to existing indications, and thus, whose categories differ
>   from and do not interact with existing categories
> 
> We can define completely independent spaces of signals by defining new
> categories to describe the new space, giving signals in one space the
> universal value for categories describing the other space.
> 
> For example, we can define
> 
>     auto
>         auto-make
>             auto-make-volvo
>             auto-make-toyota
>     color
>         color-red
>         color-green
> 
> If we have signals with the values
> 
>     auto-make-volvo & color-red
>     auto-make-volvo & color-green
>     auto-make-toyota & color-red
>     auto-make-toyota & color-green
> 
> and add them to the set of "mode-visual" signals given above, we get:
> 
>     reason & duration-normal & locale & mode-visual & auto & color
>     reason & duration-short & locale & mode-visual & auto & color
>     reason & duration-long & locale & mode-visual & auto & color
>     reason & duration & locale & mode & auto-make-volvo & color-red
>     reason & duration & locale & mode & auto-make-volvo & color-green
>     reason & duration & locale & mode & auto-make-toyota & color-red
>     reason & duration & locale & mode & auto-make-toyota & color-green
> 
> This set is a combination of chalk and cheese.  But the first
> non-universal value in the indicator will select signals of one type
> or the other.  For instance, "auto-make" will select
> 
>     reason & duration & locale & mode & auto-make-volvo & color-red
>     reason & duration & locale & mode & auto-make-volvo & color-green
>     reason & duration & locale & mode & auto-make-toyota & color-red
>     reason & duration & locale & mode & auto-make-toyota & color-green
> 
> and "duration-normal" will select
> 
>     reason & duration-normal & locale & mode-visual & auto & color
> 
> Dale
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From peter.musgrave@magorcorp.com  Tue Jun 15 12:38:59 2010
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To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [dispatch] (VIPR) - VAP in or out?
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Since the charter is just pointing to these as starting points I vote
to include VAP. I see it as a reasonable starting point.

Peter Musgrave

On Tue, Jun 15, 2010 at 1:23 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
> On Jun 2, 2010, at 12:31 PM, Marc Petit-Huguenin wrote:
>
>>>
>>> The problem statement and some possible starting points for solutions a=
re further desired in the following internet drafts which shall form the ba=
ses of the WG documents:
>>> draft-rosenberg-dispatch-vipr-overview
>>> draft-rosenberg-dispatch-vipr-reload-usage
>>> draft-rosenberg-dispatch-vipr-pvp
>>> draft-rosenberg-dispatch-vipr-sip-antispam
>>
>> VAP is not listed here. =A0Is it an oversight or is it because the WG wi=
ll work
>> only on interoperability between ViPR servers?
>
> I could go either way on VAP. On one hand, the minimal thing we need to s=
tandardize to have interoperability between domain is just the above stuff =
without VAP. On the other hand, VAP is a very simple way to modularize what=
 is happening inside a domain. VAP allows information about PSTN call to be=
 given to the server doing the VIPR stuff and allows the VIPR server to tel=
l routing engines and PBX inside the domain about new routes.
>
> I'd be interested in hearing what other thing. It's easy to add something=
 like VAP to the charter.
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From Even.roni@huawei.com  Tue Jun 15 15:25:40 2010
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Date: Wed, 16 Jun 2010 01:23:18 +0300
From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] Charter Proposal: Verification Involving PSTN	Reachability (VIPR)
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Hi Cullen,
I think that other standard body should be consulted like ITU. The reason is
that I see that one assumption is to use the PSTN numbering plan using also
the caller id. My experience is that this is not information that can be
reliable when going between PSTN service providers and it gets worse on
international calls. 
Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Tuesday, June 01, 2010 7:55 PM
> To: DISPATCH list
> Subject: [dispatch] Charter Proposal: Verification Involving PSTN
> Reachability (VIPR)
> 
> 
> I've been talking to a lot of people about the VIPR drafts  - here is a
> first cut of a proposal for a WG that could do this. I'm sure the
> charter proposal needs a bunch of work but I wanted to get the
> discussion rolling on the list.
> 
> Thanks, Cullen
> 
> (PS - this is sent in my individual contributor role. Take all my posts
> about VIPR to be in my individual role not my co-chair role)
> 
> -------------------------------------------------------
> 
> ViPR Charter Proposal (Version 0)
> 
> WG Name:  Verification Involving PSTN Reachability (VIPR)
> 
> There are two globally deployed address spaces for communications that
> more than a billion people use on a daily basis. They are phone numbers
> and DNS rooted address such as web servers and email addresses. The
> federation design of SIP is primarily designed for email style
> addresses yet a large percentage of SIP deployments primarily use phone
> numbers for identifying users. The goal of this working group is to
> allows people to use SIP to federate over the the internet while still
> using phone numbers to identify the person they wish to communicate
> with.
> 
> The VIPR WG will address this problem by developing a peer to peer
> based approach to finding SIP domains that claim to be responsible for
> a given phone number and the WG will design validation protocols to
> ensure a reasonable likelihood that a given domain actually is
> responsible for the phone number. One initial validation protocol will
> be based on a domain being able to prove it received a particular phone
> call over the PSTN based on both sides knowing the start and stop times
> of that call. Other validation schemes, such as examining fingerprints
> or watermarking of PSTN media, to show that a domain received a
> particular PSTN phone call may be considered by the working group. To
> help mitigate SPAM over SIP issues, the WG will define an token based
> authorization scheme so that domain using SIP to federate can choose to
> check that incoming SIP calls are from a domain that successfully
> validated a phone number.
> 
> The problem statement and some possible starting points for solutions
> are further desired in the following internet drafts which shall form
> the bases of the WG documents:
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
> 
> The working group will carefully coordinate with the security area, O&M
> area, as well as the appropriate RAI WG including sipcore and p2psip.
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From petithug@acm.org  Tue Jun 15 15:28:42 2010
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Subject: Re: [dispatch] (VIPR) - VAP in or out?
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 06/15/2010 10:23 AM, Cullen Jennings wrote:
> 
> On Jun 2, 2010, at 12:31 PM, Marc Petit-Huguenin wrote:
> 
>>>
>>> The problem statement and some possible starting points for solutions are further desired in the following internet drafts which shall form the bases of the WG documents:
>>> draft-rosenberg-dispatch-vipr-overview
>>> draft-rosenberg-dispatch-vipr-reload-usage
>>> draft-rosenberg-dispatch-vipr-pvp
>>> draft-rosenberg-dispatch-vipr-sip-antispam
>>
>> VAP is not listed here.  Is it an oversight or is it because the WG will work
>> only on interoperability between ViPR servers?
> 
> I could go either way on VAP. On one hand, the minimal thing we need to standardize to have interoperability between domain is just the above stuff without VAP. On the other hand, VAP is a very simple way to modularize what is happening inside a domain. VAP allows information about PSTN call to be given to the server doing the VIPR stuff and allows the VIPR server to tell routing engines and PBX inside the domain about new routes. 
> 
> I'd be interested in hearing what other thing. It's easy to add something like VAP to the charter. 
> 

I do not plan to implement VAP and so will not be able to contribute on this
protocol if the WG choose to work on it.  So for what I am concerned, VAP does
not need to be in the charter, and I think that resources will be better spent
working of stuff that directly affects interoperability (and with SIP in the
equation there will be no shortage of interop issue).

This said, the VAP I-D is a mandatory reading to understand the behavior of a
ViPR server, so there will be at least a need to split the VAP protocol from the
ViPR server behavior.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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Hi,
I read the charter and the listed drafts. I have no problem with the first
two paragraph of the charter

"There are two globally deployed address spaces for communications that more
than a billion people use on a daily basis. They are phone numbers and DNS
rooted address such as web servers and email addresses. The inter-domain
signaling design of SIP is primarily designed for email style addresses yet
a large percentage of SIP deployments mostly use phone numbers for
identifying users. The goal of this working group is to enable inter-domains
communications over the internet, using protocol such as SIP, while still
allowing people to use phone numbers to identify the person they wish to
communicate with.

The VIPR WG will address this problem by developing a peer to peer based
approach to finding domains that claim to be responsible for a given phone
number and validation protocols to ensure a reasonable likelihood that a
given domain actually is responsible for the phone number."

I have a concern about using PSTN infrastructure for the reachability. My
understanding so far was that SIP is trying to provide a new way for end to
end communication that will replace the existing circuit switch
infrastructure. This proposal says that the way to achieve end to end
connectivity requires to have an end to end PSTN frastructure.

The third paragraph is talking about validation using PSTN calls. I think
that we can look at validation of number ownership but should say that
requiring PSTN calls to do it is not the recommended approach. This will
allow managing of PSTN numbers not in the scope of PSTN infrastructure. Even
the PSTN network is using external protocol like SS7 to route calls using
databases for achieving reachability so why not say we can use similar
infrastructure that will be IP based.

I can understand this as a temporary solution but not as a standard
developed by the IETF.


 Thanks

Roni Even


From laura.liess.dt@googlemail.com  Wed Jun 16 01:11:14 2010
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References: <AANLkTimrhjA3QLBk2qN2p7mbgsELDDhb_7kUMrxvwa6U@mail.gmail.com> <4C1224FF.5050105@ericsson.com> <CD5674C3CD99574EBA7432465FC13C1B21FD73618F@DC-US1MBEX4.global.avaya.com> <4C15C46D.3010504@ericsson.com> <B12A6B04-C673-4FFC-8F9D-A211C575963C@cisco.com>
Date: Wed, 16 Jun 2010 10:11:10 +0200
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From: Laura Liess <laura.liess.dt@googlemail.com>
To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [dispatch] DISPATCH-78 topics - Alert URNs
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This would be great.
We need about 20 to 30 min for presentation and discussions on the
draft.  Weeeks ago we had ML discussions on some key issues where I
think we reached a consensus, e.g.  the definitions for the Alert-Info
URN syntax, alert-categories and alert-indications. However, my
feeling was that not everyone interested in the issue was in the ML
discussion and that we need a f2f discussion.

Thanks a lot
Laura



2010/6/15 Cullen Jennings <fluffy@cisco.com>:
>
> Let Mary and I try and to reserver 30 minutes or more for Alert-Info URN.=
 If ALert URN is a WG by that point, it can run it as a very short WG meeti=
ng and if is not a WG we can use the time to discuss how to get to being a =
WG. We still need to sort out the agenda but I don't foresee a problem with=
 this.
>
>
> On Jun 13, 2010, at 11:55 PM, Gonzalo Camarillo wrote:
>
>> Hi,
>>
>> if you need face-to-face time for this, go ahead talk to the DISPATCH
>> chairs so that you can either get a slot within the DISPATCH session or,
>> alternatively, have an ad-hoc meeting. If we managed to charter the WG
>> in time, we would have a WG meeting instead.
>>
>> Cheers,
>>
>> Gonzalo
>>
>> On 14/06/2010 5:37 AM, WORLEY, Dale R (Dale) wrote:
>>> ________________________________________
>>> From: Gonzalo Camarillo [Gonzalo.Camarillo@ericsson.com]
>>>
>>> yes, Mary's description is correct. The process is as follows. Once we
>>> agreed on the charter, I requested the IESG to review it. This charter
>>> will be discussed in the next IESG telechat (the coming Thursday),
>>> actually. Once the IESG is happy with it, the next step will be to have
>>> an IETF-wide review. After that, we will be able to charter the WG.
>>>
>>> In any case, as we have said a number of times, proponents of a
>>> particular piece of work should not stop working on it while we charter
>>> their WG. They are encouraged to continue making progress in parallel
>>> with the chartering process.
>>> ________________________________________
>>>
>>> That is all fine, but what are the implications regarding getting a tim=
e slot at Maastricht? =A0There seem to be two choices: =A0a slot within Dis=
patch, or a separately scheduled Salud session. =A0What we do not want to g=
et stuck with is not being able to get time in Dispatch (because a charter =
has been proposed) and not being able to get a separate session (because th=
e charter hasn't gone through all the approval steps), as that would preven=
t any session at =A0Maastricht.
>>>
>>> Dale
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From peter.musgrave@magorcorp.com  Wed Jun 16 04:53:16 2010
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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Hi,

I think the use of PSTN reachability is central to ViPR and it's
value. I agree there will be other ways of authenticating and that
over time they may displace ViPR but I think PSTN reachability will be
around for a long time. I think it is the lowest-common denominator
and avoids things like trying to interface to databases of PSTN
numbers and all the assorted standards and security issues that
entails.

I see this as a temporary solution in the same way that NAT traversal
is a temporary solution :-)

Perhaps we should make it clear in the charter that the work requires
that a node using ViPR have the ability to make and receive PSTN calls
(or interface to something that does)?

I think general authentication without using PSTN reachability is a
great idea - but I think that is a far larger and more complex issue.
IIUC ViPR exists to circumvent that issue and provide a shorter term,
standards-based, pragmatic solution. I think the more general problem
would need a charter unto itself.

Peter Musgrave

On Wed, Jun 16, 2010 at 2:22 AM, Roni Even <Even.roni@huawei.com> wrote:
> Hi,
> I read the charter and the listed drafts. I have no problem with the firs=
t
> two paragraph of the charter
>
> "There are two globally deployed address spaces for communications that m=
ore
> than a billion people use on a daily basis. They are phone numbers and DN=
S
> rooted address such as web servers and email addresses. The inter-domain
> signaling design of SIP is primarily designed for email style addresses y=
et
> a large percentage of SIP deployments mostly use phone numbers for
> identifying users. The goal of this working group is to enable inter-doma=
ins
> communications over the internet, using protocol such as SIP, while still
> allowing people to use phone numbers to identify the person they wish to
> communicate with.
>
> The VIPR WG will address this problem by developing a peer to peer based
> approach to finding domains that claim to be responsible for a given phon=
e
> number and validation protocols to ensure a reasonable likelihood that a
> given domain actually is responsible for the phone number."
>
> I have a concern about using PSTN infrastructure for the reachability. My
> understanding so far was that SIP is trying to provide a new way for end =
to
> end communication that will replace the existing circuit switch
> infrastructure. This proposal says that the way to achieve end to end
> connectivity requires to have an end to end PSTN frastructure.
>
> The third paragraph is talking about validation using PSTN calls. I think
> that we can look at validation of number ownership but should say that
> requiring PSTN calls to do it is not the recommended approach. This will
> allow managing of PSTN numbers not in the scope of PSTN infrastructure. E=
ven
> the PSTN network is using external protocol like SS7 to route calls using
> databases for achieving reachability so why not say we can use similar
> infrastructure that will be IP based.
>
> I can understand this as a temporary solution but not as a standard
> developed by the IETF.
>
>
> =A0Thanks
>
> Roni Even
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From pkyzivat@cisco.com  Wed Jun 16 05:51:50 2010
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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Roni Even wrote:

> I have a concern about using PSTN infrastructure for the reachability. My
> understanding so far was that SIP is trying to provide a new way for end to
> end communication that will replace the existing circuit switch
> infrastructure. This proposal says that the way to achieve end to end
> connectivity requires to have an end to end PSTN frastructure.
> 
> The third paragraph is talking about validation using PSTN calls. I think
> that we can look at validation of number ownership but should say that
> requiring PSTN calls to do it is not the recommended approach. This will
> allow managing of PSTN numbers not in the scope of PSTN infrastructure. Even
> the PSTN network is using external protocol like SS7 to route calls using
> databases for achieving reachability so why not say we can use similar
> infrastructure that will be IP based.
> 
> I can understand this as a temporary solution but not as a standard
> developed by the IETF.

I have had similar concerns for some time - this strategy is not an "end 
game".

BUT, at the moment there is nothing better for most phone numbers.
That will probably be true for quite some time. So I think it must be 
*part* of the solution. It is a way to bootstrap the process.

Note that this is compatible with using SIP for "PSTN" calls. Many SPs 
now have SIP Trunks, and calls over these can be used for ViPR validation.

Eventually there should be some other mechanism that doesn't depend on 
PSTN calls. Maybe ViPR will motivate the deployment of Public ENUM. Or 
maybe there will be something else.  But in any case having *something* 
in place should kickstart the process.

	Thanks,
	Paul

From laura.liess.dt@googlemail.com  Wed Jun 16 06:12:09 2010
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Date: Wed, 16 Jun 2010 15:12:05 +0200
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From: Laura Liess <laura.liess.dt@googlemail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "WORLEY, Dale R (Dale)" <dworley@avaya.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Dale, Paul,

what about trying to combine Dale's conceptual framework (which also
seems to me to solve the semantic-non semantic discussion) and the
syntax Paul and John and I agreed upon
http://www.ietf.org/mail-archive/web/dispatch/current/msg01618.html
back in April ?  I think they could fit together, or do you see any
issues where they don't?

Laura

2010/6/15 Paul Kyzivat <pkyzivat@cisco.com>:
> Very interesting!
>
> Without getting into syntax specifics, I think I like this - it provides =
a
> much cleaner conceptual framework. The idea of the specifier providing a
> priority seems to resolve some loose ends.
>
> =A0 =A0 =A0 =A0Thanks,
> =A0 =A0 =A0 =A0Paul
>
> WORLEY, Dale R (Dale) wrote:
>>
>> [As I am the chair-elect for this topic, I now have to identify that I
>> am proposing this as an individual contributor-elect.]
>>
>> This is a conceptual proposal for for specifying Alert-Info URNs. =A0The
>> problem that I am attacking is extensibility -- because the practical
>> application of the Alert-Info URNs will be as successors of the
>> ring tone and ringback tone implementations of innumerable existing
>> telephone systems, we expect the uses of Alert-Info URNs to be subject
>> to much extension. =A0The primary problem is to provide a framework
>> which allows such extensions to be implemented without damaging
>> interoperability between systems whose Alert-Info usage has not been
>> coordinated a priori. =A0A secondary problem is to ensure that the
>> framework can be implemented efficiently in practical situations.
>>
>> The basis of this proposal has been stolen from previous contributors
>> to the Alert-Info work.
>>
>> Since this is a conceptual proposal, I am not particular about the
>> syntax to be used -- the syntax shown here is just to illustrate the
>> concepts. =A0What I am proposing is a system of semantics; many variant
>> syntaxes can be used to implement it with differing tradeoffs.
>>
>> * Terminology
>>
>> We say that a "specifier" sends an "indication" (a URN in an
>> Alert-Info header) to a "renderer" which then "renders" a "signal"
>> based on the indication to a human user. =A0A "category" is a
>> characteristic whose "values" can be used to classify indications.
>>
>> * Requirements and non-requirements
>>
>> Requirement A: =A0Indications must be able to represent a wide variety
>> of signals, which have many largely-orthogonal characteristics. =A0(See
>> draft-liess-dispatch-alert-info-urns-00 section section 2 for a
>> partial list, and Paul Kyzivat's message for another partial list.)
>>
>> Requirement B: =A0Indications include subsets which have distinctly
>> different semantics, that is, they form disjoint "value spaces". =A0For
>> example, some indications should describe the semantics of the
>> signaling situation whereas others should describe the audio
>> characteristics of the signal. =A0This implies that there is no single
>> set of categories that can be used as independent coordinates of the
>> value-space of indications.
>>
>> Requirement C: =A0The set of indications must be able to support
>> extensibility by a wide variety of organizations that are not
>> coordinated with each other. =A0Extensions must be able to:
>>
>> - add further values to any existing category
>>
>> - add further categories that are orthogonal to existing categories
>>
>> - semantically subdivide the meaning provided by any existing
>> =A0indication
>>
>> - add further "value spaces" of indications whose semantics are not
>> =A0related to existing indications, and thus, whose categories differ
>> =A0from and do not interact with existing categories
>>
>> Requirement D: =A0An indication is transmitted from a specifier to a
>> renderer, which must base its rendering decision only on the
>> indication. =A0In particular, there is no multi-message negotiation
>> process or carrying of context from one indication to the next.
>>
>> Requirement E: =A0The renderer may be customized in ways that limit the
>> set of signals that it can render, or it may be provided with a set of
>> signals that have uncommon semantics. =A0(The canonical example is a UA
>> for the hearing-impaired.) =A0(By Requirement D, the renderer has no way
>> of transmitting this fact to the specifier.)
>>
>> Requirement F: =A0If the specifier and the renderer have designs that
>> are properly coordinated, the indications must be able to reliably
>> carry all extensions that are supported in the coordinated designs.
>>
>> Requirement G: =A0In any situation, the renderer must be able to perform
>> close to the best possible rendering that it could do even the
>> specifier had specific knowledge of the renderer's capabilities.
>>
>> Non-Requirement H: =A0In any situation other than that of Requirement F,
>> we do not require the render to perform the best possible rendering.
>>
>> * Categories, values, and indicators
>>
>> There are a theoretically infinite number of categories. =A0Each
>> category is named by a word, such as:
>>
>> =A0 =A0reason
>> =A0 =A0priority
>> =A0 =A0source
>> =A0 =A0duration
>>
>> The values of each category conceptually form a tree. =A0The root of the
>> tree is the universal, or most general, value. =A0It is denoted by the
>> the category name, and represents the universe of all possibilities.
>> Each further subdivision of a value is specified by adding a hyphen
>> and another word. =A0Thus, within the category "state" we have values:
>>
>> =A0 =A0state
>> =A0 =A0state-unknown
>> =A0 =A0state-waiting
>> =A0 =A0state-on.call
>> =A0 =A0state-away
>>
>> (Here we pretend '.' is a letter.)
>>
>> The first of these values specifies no specific information. =A0The
>> following values specify subdivisions of the semantic space. =A0The
>> value "state-waiting" can be further subdivided based on the
>> importance of the call being waited upon:
>>
>> =A0 =A0state-waiting-standard
>> =A0 =A0state-waiting-vip
>>
>> An indication is assembled by providing a sequence of values (which
>> implicitly specify their categories), separated by colons, in order of
>> importance. =A0That is, the first value is the one that should be
>> rendered as accurately as possible; within that constraint, the second
>> value should be rendered as accurately as possible; etc.
>>
>> For example, we can assemble the following indications:
>>
>> =A0 =A0an internal, priority call
>> =A0 =A0 =A0 =A0source-internal:priority-high
>>
>> =A0 =A0priority call waiting
>> =A0 =A0 =A0 =A0reason-call.waiting:priority-high
>>
>> =A0 =A0short Albanian auto-callback
>> =A0 =A0 =A0 =A0reason-auto.callback:duration-short:locale-al
>>
>> (The specific examples are taken from
>> draft-liess-dispatch-alert-info-urns-00 section 4.5, the specific
>> categories and values are adapted from Paul Kyzivat's message.)
>>
>> All of these indications are easily extensible. =A0For instance, if
>> Albania has two service providers with different signal tones, we
>> could specialize the last indication into:
>>
>> =A0 =A0short Albanian auto-callback, service provider A
>> =A0 =A0 =A0 =A0reason-auto.callback:duration-short:locale-al-sp.A
>>
>> =A0 =A0short Albanian auto-callback, service provider B
>> =A0 =A0 =A0 =A0reason-auto.callback:duration-short:locale-al-sp.B
>>
>> Due to the rule that the earlier values in an indicator have more
>> influence than latter values, there is a subtle difference between:
>>
>> =A0 =A0source-internal:priority-high
>> =A0 =A0priority-high:source-internal
>>
>> in regard to which aspect of the indication it is more important to
>> render.
>>
>> * Selecting the signal based on an indication
>>
>> Having defined a universe of indications with infinite extensibility,
>> we now need to show that a practical device can interpret indications
>> to select from a limited set of signals. =A0To do this, I will describe
>> a very general algorithm for doing this selection in all possible
>> situations. =A0Note that in practical cases (where all the signals that
>> can be rendered can be described by a small number of values of a
>> small number of categories) the algorithm will operate in a reasonably
>> rapid and simple way.
>>
>> To start with, we assume that the renderer possesses a set of
>> available signals. =A0This set will naturally be partially determined by
>> the renderer itself, but the set may also be affected by the
>> configuration of the renderer (e.g., hearing-impaired mode vs. hearing
>> mode) and the context of the indication (e.g., ring tone vs. ringback
>> tone situations). =A0The set may be finite or infinite. =A0Of course, if
>> the set is infinite, it can't be represented literally, but the
>> renderer necessarily has a method of representing the set and its
>> possible subsets. =A0A finite set of available signals can be enumerated
>> directly, or represented in some other way.
>>
>> Each available signal is described by the values of a set of
>> categories. =A0We assume that the renderer only knows the values for a
>> finite set of categories; for all other categories, all available
>> signals implicitly have the "universal", root value.
>>
>> Informally, the algorithm proceeds as follows:
>>
>> =A0 =A0Load a set of "acceptable signals" with all available signals.
>>
>> =A0 =A0Process the words in the values in the indication from left to ri=
ght.
>> =A0 =A0As each additional word is considered, consider the portion of th=
e
>> =A0 =A0value that has been examined, and remove from the acceptable set =
all
>> =A0 =A0signals which are inconsistent with that value, in that their val=
ue
>> =A0 =A0for that category is not equal to or below the indication's value=
 (as
>> =A0 =A0processed so far).
>>
>> =A0 =A0 =A0 =A0If after processing a word, one or more signals remain in=
 the
>> =A0 =A0 =A0 =A0acceptable set, continue processing the next word.
>>
>> =A0 =A0 =A0 =A0If after processing a word, no signals remain in the
>> =A0 =A0 =A0 =A0acceptable set, restore the previous state of the accepta=
ble
>> =A0 =A0 =A0 =A0set and continue with the next value in the indication.
>>
>> =A0 =A0At the end of processing, the acceptable set will contain one or =
more
>> =A0 =A0signals. =A0Choose from among them according to some sort of prio=
rity
>> =A0 =A0system.
>>
>> More formally, an indication is processed by:
>>
>> =A0 =A0 =A0initialize the acceptable set of signals with all available s=
ignals
>>
>> =A0 =A0 =A0for each value in the indication (going left to right):
>>
>> =A0 =A0 =A0 =A0 =A0for each word in the value (going left to right, whic=
h is down
>> =A0 =A0 =A0 =A0 =A0the tree of values):
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0if there are acceptable signals which have a =
value of the
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0category which is equal to or below the indic=
ation value as
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0seen so far:
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0delete all acceptable signals which d=
o not have such a
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0value, and continue processing this v=
alue
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0else
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0cease processing this value and conti=
nue with the next
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0value
>>
>> =A0 =A0 =A0the acceptable set will contain at least one signal. =A0Choos=
e from
>> =A0 =A0 =A0among the acceptable signals via some priority scheme.
>>
>> Let me work through an example.
>>
>> Let us suppose that the renderer implements three categories:
>>
>> =A0 =A0reason
>> =A0 =A0 =A0 =A0reason-normal
>> =A0 =A0 =A0 =A0reason-call.waiting
>> =A0 =A0 =A0 =A0reason-forward
>> =A0 =A0 =A0 =A0reason-auto.callback
>> =A0 =A0 =A0 =A0reason-recall.hold
>> =A0 =A0 =A0 =A0reason-recall.transfer
>>
>> =A0 =A0duration
>> =A0 =A0 =A0 =A0duration-normal
>> =A0 =A0 =A0 =A0duration-short
>> =A0 =A0 =A0 =A0duration-long
>>
>> =A0 =A0locale
>> =A0 =A0 =A0 =A0locale-fm
>> =A0 =A0 =A0 =A0locale-al
>> =A0 =A0 =A0 =A0 =A0 =A0locale-al-sp.A
>> =A0 =A0 =A0 =A0 =A0 =A0locale-al-sp.B
>>
>> Let us suppose that the renderer has signals for all combinations of
>> the leaf values of these categories, except that for historical
>> reasons if "reason" has a non-universal value, "default" must be
>> universal, and vice-versa; the signals (copied from the PSTN service
>> providers) do not allow signaling a non-standard reason with a
>> non-standard duration. =A0So the allowed combinations of these two
>> values are:
>>
>> =A0 =A0reason-normal & duration
>> =A0 =A0reason-call.waiting & duration
>> =A0 =A0reason-forward & duration
>> =A0 =A0reason-auto.callback & duration
>> =A0 =A0reason-recall.hold & duration
>> =A0 =A0reason-recall.transfer & duration
>> =A0 =A0reason & duration-normal
>> =A0 =A0reason & duration-short
>> =A0 =A0reason & duration-long
>>
>> There are signals that combine each of these with "locale-fm",
>> "locale-al-sp.A", and "locale-al-sp.B".
>>
>> To process the indication
>> "reason-auto.callback:duration-short:locale-al-sp.A", the renderer
>> first loads the acceptable signal set with all 27 of its signals, and
>> then examines the indication word by word:
>>
>> - "reason"
>>
>> Removes no signals from the set.
>>
>> - "reason-auto.callback"
>>
>> Reduces the set to
>>
>> =A0 =A0reason-auto.callback & duration & locale-fm
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.A
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.B
>>
>> - "duration"
>>
>> Removes no signals from the set.
>>
>> - "duration-short"
>>
>> Since no signals in the set have a value at or under this value, no
>> change is made and the processing of this value ends.
>>
>> - "locale"
>>
>> Removes no signals from the set.
>>
>> - "locale-al"
>>
>> Reduces the set to
>>
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.A
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.B
>>
>> - "locale-al-sp.A"
>>
>> Reduces the set to
>>
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.A
>>
>> The end of the indication is reached, and since only one signal remains
>> in the set, it is rendered.
>>
>> Now, if the indication was less specific,
>> "reason-auto.callback:duration-short:locale-al", at the end the
>> acceptable set would be
>>
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.A
>> =A0 =A0reason-auto.callback & duration & locale-al-sp.B
>>
>> and the renderer would have to choose between these two signals in
>> some way.
>>
>> Let us suppose that the renderer implemented a further category:
>>
>> =A0 =A0mode
>> =A0 =A0 =A0 =A0mode-visual
>> =A0 =A0 =A0 =A0mode-audio
>>
>> and applied the value "mode-audio" to all of the above signals, but
>> only added the following signals with "mode-visual":
>>
>> =A0 =A0reason & duration-normal & locale & mode-visual
>> =A0 =A0reason & duration-short & locale & mode-visual
>> =A0 =A0reason & duration-long & locale & mode-visual
>>
>> That is, the only variation provided is in "duration".
>>
>> Then the indication "reason-auto.callback:duration-short:locale-al"
>> would generate the same acceptable set as above, that is, there would
>> be no "mode-visual" signals. =A0The indication
>> "mode-visual:reason-auto.callback:duration-short:locale-al" would
>> generate the acceptable set:
>>
>> =A0 =A0reason & duration-short & locale & mode-visual
>>
>> This is because the indication implies that matching "mode" is more
>> important than matching "reason" and "duration". =A0Once the selection
>> for "mode-visual" has been done, no further selection can be done on
>> "reason", but "duration-short" can be acted upon.
>>
>> In practice, for use with a hearing-impaired user, the initial set of
>> signals would be restricted to those with "mode-visual". =A0In this
>> example, we have assumed that the user considers "mode-visual" and
>> "mode-audio" to be equally acceptable.
>>
>> If we configured the renderer to only use "mode-visual" signals, then
>> the indication "reason-auto.callback:duration-short:locale-al" would
>> select only the signal
>>
>> =A0 =A0reason & duration-short & locale & mode-visual
>>
>> In this configuration, the indication
>> "mode-audio:reason-auto.callback:duration-short:locale-al" would also
>> select that signal, because the value "mode-audio" cannot be acted
>> upon within that set of available signals. =A0It is effectively ignored.
>>
>> * Extensibility
>>
>> Let us now see how this scheme supports the various types of
>> extensibility listed in Requirement C.
>>
>> - add further values to any existing category
>>
>> We could define an additional value such as
>>
>> =A0 =A0reason-avaya.special
>>
>> If the renderer had any signals that had this value, they would be
>> selected by an indicator containing "reason-avaya.special". =A0If the
>> renderer has no such signals, then in an indicator this value acts the
>> same as "reason", that is, it has no effect.
>>
>> - add further categories that are orthogonal to existing categories
>>
>> We could define an additional category with values:
>>
>> =A0 =A0avaya
>> =A0 =A0 =A0 =A0avaya-a
>> =A0 =A0 =A0 =A0avaya-b
>> =A0 =A0 =A0 =A0avaya-c
>>
>> If a renderer does not know of this category, all of its signals
>> implicitly have the value "avaya". =A0Because of this, when such a
>> renderer processes a value of this category in an indicator, it never
>> changes the available set.
>>
>> - semantically subdivide the meaning provided by any existing
>> =A0indication
>>
>> We could define subvalues of existing values, such as:
>>
>> =A0 =A0state
>> =A0 =A0 =A0 =A0state-waiting
>> =A0 =A0 =A0 =A0 =A0 =A0state-waiting-avaya.standard
>> =A0 =A0 =A0 =A0 =A0 =A0state-waiting-avaya.vip
>>
>> If the renderer has two signals that differ only in the values
>> "state-waiting-avaya.standard" and "state-waiting-avaya.vip", any
>> indicator that does not know of this subdivision will never separate
>> the two signals; they will both remain in the acceptable set or both
>> be excluded from it. =A0So the renderer needs a policy for choosing one
>> over the other -- In this case, clearly "state-waiting-avaya.standard"
>> is to be preferred.
>>
>> However, if the indicator gives one of the two subvalues, the renderer
>> will select the appropriate signal.
>>
>> If the indicator gives one of the two subvalues but the renderer only
>> knows of "state-waiting", the "avaya.standard" or "avaya.vip" part of
>> the value will never affect the acceptable set (since to select based
>> on it would make the acceptable set empty). =A0Processing of
>> "state-waiting-avaya.standard" would have the same effect as
>> "state-waiting".
>>
>> - add further "value spaces" of indications whose semantics are not
>> =A0related to existing indications, and thus, whose categories differ
>> =A0from and do not interact with existing categories
>>
>> We can define completely independent spaces of signals by defining new
>> categories to describe the new space, giving signals in one space the
>> universal value for categories describing the other space.
>>
>> For example, we can define
>>
>> =A0 =A0auto
>> =A0 =A0 =A0 =A0auto-make
>> =A0 =A0 =A0 =A0 =A0 =A0auto-make-volvo
>> =A0 =A0 =A0 =A0 =A0 =A0auto-make-toyota
>> =A0 =A0color
>> =A0 =A0 =A0 =A0color-red
>> =A0 =A0 =A0 =A0color-green
>>
>> If we have signals with the values
>>
>> =A0 =A0auto-make-volvo & color-red
>> =A0 =A0auto-make-volvo & color-green
>> =A0 =A0auto-make-toyota & color-red
>> =A0 =A0auto-make-toyota & color-green
>>
>> and add them to the set of "mode-visual" signals given above, we get:
>>
>> =A0 =A0reason & duration-normal & locale & mode-visual & auto & color
>> =A0 =A0reason & duration-short & locale & mode-visual & auto & color
>> =A0 =A0reason & duration-long & locale & mode-visual & auto & color
>> =A0 =A0reason & duration & locale & mode & auto-make-volvo & color-red
>> =A0 =A0reason & duration & locale & mode & auto-make-volvo & color-green
>> =A0 =A0reason & duration & locale & mode & auto-make-toyota & color-red
>> =A0 =A0reason & duration & locale & mode & auto-make-toyota & color-gree=
n
>>
>> This set is a combination of chalk and cheese. =A0But the first
>> non-universal value in the indicator will select signals of one type
>> or the other. =A0For instance, "auto-make" will select
>>
>> =A0 =A0reason & duration & locale & mode & auto-make-volvo & color-red
>> =A0 =A0reason & duration & locale & mode & auto-make-volvo & color-green
>> =A0 =A0reason & duration & locale & mode & auto-make-toyota & color-red
>> =A0 =A0reason & duration & locale & mode & auto-make-toyota & color-gree=
n
>>
>> and "duration-normal" will select
>>
>> =A0 =A0reason & duration-normal & locale & mode-visual & auto & color
>>
>> Dale
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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Date: Wed, 16 Jun 2010 09:39:05 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Laura Liess wrote:
> Dale, Paul,
> 
> what about trying to combine Dale's conceptual framework (which also
> seems to me to solve the semantic-non semantic discussion) and the
> syntax Paul and John and I agreed upon
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01618.html
> back in April ?  I think they could fit together, or do you see any
> issues where they don't?

I was trying to avoid syntax issues for the moment, in order to focus on 
the essence of Dale's proposal. But if we generally like where that is 
going then eventually it will come back to syntax.

IIUC, in what Dale has proposed the "indication" includes a prioritized 
list of capability-values. If we followed the syntax of the current 
draft, then the prioritization would probably come by considering the 
order of URNs in the Alert-Info header to be significant.

I think that could work. It might even be possible to generalize the 
rule to the treatment of all sorts of URIs in the Alert-Info header, not 
just this particular class of URNs. The rule would be something like:

- process URIs from the A-I header in order. For each:
   - if it is understood, and is acceptable according to local policy:
     - if it can be used in conjunction with already accepted URIs:
       - then accept this one as well
       - else discard it
- once that is complete, if nothing has been accepted,
   or if some aspect of alerting as not been addressed by
   what has been accepted, then fill in suitable defaults.
- use the accepted URIs, plus defaults, to do the alerting.

The above is a higher level "meta-policy" for alerting, within which the 
selection and combination rules that Dale spells out can be applied.

One thing that *isn't* covered by Dale, or the above, is what to do if 
the "renderer" disagrees with the "specifier" about the priorities. I 
don't (yet) have an opinion if that is a problem that needs solving.

	Thanks,
	Paul

> Laura
> 
> 2010/6/15 Paul Kyzivat <pkyzivat@cisco.com>:
>> Very interesting!
>>
>> Without getting into syntax specifics, I think I like this - it provides a
>> much cleaner conceptual framework. The idea of the specifier providing a
>> priority seems to resolve some loose ends.
>>
>>        Thanks,
>>        Paul
>>
>> WORLEY, Dale R (Dale) wrote:
>>> [As I am the chair-elect for this topic, I now have to identify that I
>>> am proposing this as an individual contributor-elect.]
>>>
>>> This is a conceptual proposal for for specifying Alert-Info URNs.  The
>>> problem that I am attacking is extensibility -- because the practical
>>> application of the Alert-Info URNs will be as successors of the
>>> ring tone and ringback tone implementations of innumerable existing
>>> telephone systems, we expect the uses of Alert-Info URNs to be subject
>>> to much extension.  The primary problem is to provide a framework
>>> which allows such extensions to be implemented without damaging
>>> interoperability between systems whose Alert-Info usage has not been
>>> coordinated a priori.  A secondary problem is to ensure that the
>>> framework can be implemented efficiently in practical situations.
>>>
>>> The basis of this proposal has been stolen from previous contributors
>>> to the Alert-Info work.
>>>
>>> Since this is a conceptual proposal, I am not particular about the
>>> syntax to be used -- the syntax shown here is just to illustrate the
>>> concepts.  What I am proposing is a system of semantics; many variant
>>> syntaxes can be used to implement it with differing tradeoffs.
>>>
>>> * Terminology
>>>
>>> We say that a "specifier" sends an "indication" (a URN in an
>>> Alert-Info header) to a "renderer" which then "renders" a "signal"
>>> based on the indication to a human user.  A "category" is a
>>> characteristic whose "values" can be used to classify indications.
>>>
>>> * Requirements and non-requirements
>>>
>>> Requirement A:  Indications must be able to represent a wide variety
>>> of signals, which have many largely-orthogonal characteristics.  (See
>>> draft-liess-dispatch-alert-info-urns-00 section section 2 for a
>>> partial list, and Paul Kyzivat's message for another partial list.)
>>>
>>> Requirement B:  Indications include subsets which have distinctly
>>> different semantics, that is, they form disjoint "value spaces".  For
>>> example, some indications should describe the semantics of the
>>> signaling situation whereas others should describe the audio
>>> characteristics of the signal.  This implies that there is no single
>>> set of categories that can be used as independent coordinates of the
>>> value-space of indications.
>>>
>>> Requirement C:  The set of indications must be able to support
>>> extensibility by a wide variety of organizations that are not
>>> coordinated with each other.  Extensions must be able to:
>>>
>>> - add further values to any existing category
>>>
>>> - add further categories that are orthogonal to existing categories
>>>
>>> - semantically subdivide the meaning provided by any existing
>>>  indication
>>>
>>> - add further "value spaces" of indications whose semantics are not
>>>  related to existing indications, and thus, whose categories differ
>>>  from and do not interact with existing categories
>>>
>>> Requirement D:  An indication is transmitted from a specifier to a
>>> renderer, which must base its rendering decision only on the
>>> indication.  In particular, there is no multi-message negotiation
>>> process or carrying of context from one indication to the next.
>>>
>>> Requirement E:  The renderer may be customized in ways that limit the
>>> set of signals that it can render, or it may be provided with a set of
>>> signals that have uncommon semantics.  (The canonical example is a UA
>>> for the hearing-impaired.)  (By Requirement D, the renderer has no way
>>> of transmitting this fact to the specifier.)
>>>
>>> Requirement F:  If the specifier and the renderer have designs that
>>> are properly coordinated, the indications must be able to reliably
>>> carry all extensions that are supported in the coordinated designs.
>>>
>>> Requirement G:  In any situation, the renderer must be able to perform
>>> close to the best possible rendering that it could do even the
>>> specifier had specific knowledge of the renderer's capabilities.
>>>
>>> Non-Requirement H:  In any situation other than that of Requirement F,
>>> we do not require the render to perform the best possible rendering.
>>>
>>> * Categories, values, and indicators
>>>
>>> There are a theoretically infinite number of categories.  Each
>>> category is named by a word, such as:
>>>
>>>    reason
>>>    priority
>>>    source
>>>    duration
>>>
>>> The values of each category conceptually form a tree.  The root of the
>>> tree is the universal, or most general, value.  It is denoted by the
>>> the category name, and represents the universe of all possibilities.
>>> Each further subdivision of a value is specified by adding a hyphen
>>> and another word.  Thus, within the category "state" we have values:
>>>
>>>    state
>>>    state-unknown
>>>    state-waiting
>>>    state-on.call
>>>    state-away
>>>
>>> (Here we pretend '.' is a letter.)
>>>
>>> The first of these values specifies no specific information.  The
>>> following values specify subdivisions of the semantic space.  The
>>> value "state-waiting" can be further subdivided based on the
>>> importance of the call being waited upon:
>>>
>>>    state-waiting-standard
>>>    state-waiting-vip
>>>
>>> An indication is assembled by providing a sequence of values (which
>>> implicitly specify their categories), separated by colons, in order of
>>> importance.  That is, the first value is the one that should be
>>> rendered as accurately as possible; within that constraint, the second
>>> value should be rendered as accurately as possible; etc.
>>>
>>> For example, we can assemble the following indications:
>>>
>>>    an internal, priority call
>>>        source-internal:priority-high
>>>
>>>    priority call waiting
>>>        reason-call.waiting:priority-high
>>>
>>>    short Albanian auto-callback
>>>        reason-auto.callback:duration-short:locale-al
>>>
>>> (The specific examples are taken from
>>> draft-liess-dispatch-alert-info-urns-00 section 4.5, the specific
>>> categories and values are adapted from Paul Kyzivat's message.)
>>>
>>> All of these indications are easily extensible.  For instance, if
>>> Albania has two service providers with different signal tones, we
>>> could specialize the last indication into:
>>>
>>>    short Albanian auto-callback, service provider A
>>>        reason-auto.callback:duration-short:locale-al-sp.A
>>>
>>>    short Albanian auto-callback, service provider B
>>>        reason-auto.callback:duration-short:locale-al-sp.B
>>>
>>> Due to the rule that the earlier values in an indicator have more
>>> influence than latter values, there is a subtle difference between:
>>>
>>>    source-internal:priority-high
>>>    priority-high:source-internal
>>>
>>> in regard to which aspect of the indication it is more important to
>>> render.
>>>
>>> * Selecting the signal based on an indication
>>>
>>> Having defined a universe of indications with infinite extensibility,
>>> we now need to show that a practical device can interpret indications
>>> to select from a limited set of signals.  To do this, I will describe
>>> a very general algorithm for doing this selection in all possible
>>> situations.  Note that in practical cases (where all the signals that
>>> can be rendered can be described by a small number of values of a
>>> small number of categories) the algorithm will operate in a reasonably
>>> rapid and simple way.
>>>
>>> To start with, we assume that the renderer possesses a set of
>>> available signals.  This set will naturally be partially determined by
>>> the renderer itself, but the set may also be affected by the
>>> configuration of the renderer (e.g., hearing-impaired mode vs. hearing
>>> mode) and the context of the indication (e.g., ring tone vs. ringback
>>> tone situations).  The set may be finite or infinite.  Of course, if
>>> the set is infinite, it can't be represented literally, but the
>>> renderer necessarily has a method of representing the set and its
>>> possible subsets.  A finite set of available signals can be enumerated
>>> directly, or represented in some other way.
>>>
>>> Each available signal is described by the values of a set of
>>> categories.  We assume that the renderer only knows the values for a
>>> finite set of categories; for all other categories, all available
>>> signals implicitly have the "universal", root value.
>>>
>>> Informally, the algorithm proceeds as follows:
>>>
>>>    Load a set of "acceptable signals" with all available signals.
>>>
>>>    Process the words in the values in the indication from left to right.
>>>    As each additional word is considered, consider the portion of the
>>>    value that has been examined, and remove from the acceptable set all
>>>    signals which are inconsistent with that value, in that their value
>>>    for that category is not equal to or below the indication's value (as
>>>    processed so far).
>>>
>>>        If after processing a word, one or more signals remain in the
>>>        acceptable set, continue processing the next word.
>>>
>>>        If after processing a word, no signals remain in the
>>>        acceptable set, restore the previous state of the acceptable
>>>        set and continue with the next value in the indication.
>>>
>>>    At the end of processing, the acceptable set will contain one or more
>>>    signals.  Choose from among them according to some sort of priority
>>>    system.
>>>
>>> More formally, an indication is processed by:
>>>
>>>      initialize the acceptable set of signals with all available signals
>>>
>>>      for each value in the indication (going left to right):
>>>
>>>          for each word in the value (going left to right, which is down
>>>          the tree of values):
>>>
>>>              if there are acceptable signals which have a value of the
>>>              category which is equal to or below the indication value as
>>>              seen so far:
>>>
>>>                  delete all acceptable signals which do not have such a
>>>                  value, and continue processing this value
>>>
>>>              else
>>>
>>>                  cease processing this value and continue with the next
>>>                  value
>>>
>>>      the acceptable set will contain at least one signal.  Choose from
>>>      among the acceptable signals via some priority scheme.
>>>
>>> Let me work through an example.
>>>
>>> Let us suppose that the renderer implements three categories:
>>>
>>>    reason
>>>        reason-normal
>>>        reason-call.waiting
>>>        reason-forward
>>>        reason-auto.callback
>>>        reason-recall.hold
>>>        reason-recall.transfer
>>>
>>>    duration
>>>        duration-normal
>>>        duration-short
>>>        duration-long
>>>
>>>    locale
>>>        locale-fm
>>>        locale-al
>>>            locale-al-sp.A
>>>            locale-al-sp.B
>>>
>>> Let us suppose that the renderer has signals for all combinations of
>>> the leaf values of these categories, except that for historical
>>> reasons if "reason" has a non-universal value, "default" must be
>>> universal, and vice-versa; the signals (copied from the PSTN service
>>> providers) do not allow signaling a non-standard reason with a
>>> non-standard duration.  So the allowed combinations of these two
>>> values are:
>>>
>>>    reason-normal & duration
>>>    reason-call.waiting & duration
>>>    reason-forward & duration
>>>    reason-auto.callback & duration
>>>    reason-recall.hold & duration
>>>    reason-recall.transfer & duration
>>>    reason & duration-normal
>>>    reason & duration-short
>>>    reason & duration-long
>>>
>>> There are signals that combine each of these with "locale-fm",
>>> "locale-al-sp.A", and "locale-al-sp.B".
>>>
>>> To process the indication
>>> "reason-auto.callback:duration-short:locale-al-sp.A", the renderer
>>> first loads the acceptable signal set with all 27 of its signals, and
>>> then examines the indication word by word:
>>>
>>> - "reason"
>>>
>>> Removes no signals from the set.
>>>
>>> - "reason-auto.callback"
>>>
>>> Reduces the set to
>>>
>>>    reason-auto.callback & duration & locale-fm
>>>    reason-auto.callback & duration & locale-al-sp.A
>>>    reason-auto.callback & duration & locale-al-sp.B
>>>
>>> - "duration"
>>>
>>> Removes no signals from the set.
>>>
>>> - "duration-short"
>>>
>>> Since no signals in the set have a value at or under this value, no
>>> change is made and the processing of this value ends.
>>>
>>> - "locale"
>>>
>>> Removes no signals from the set.
>>>
>>> - "locale-al"
>>>
>>> Reduces the set to
>>>
>>>    reason-auto.callback & duration & locale-al-sp.A
>>>    reason-auto.callback & duration & locale-al-sp.B
>>>
>>> - "locale-al-sp.A"
>>>
>>> Reduces the set to
>>>
>>>    reason-auto.callback & duration & locale-al-sp.A
>>>
>>> The end of the indication is reached, and since only one signal remains
>>> in the set, it is rendered.
>>>
>>> Now, if the indication was less specific,
>>> "reason-auto.callback:duration-short:locale-al", at the end the
>>> acceptable set would be
>>>
>>>    reason-auto.callback & duration & locale-al-sp.A
>>>    reason-auto.callback & duration & locale-al-sp.B
>>>
>>> and the renderer would have to choose between these two signals in
>>> some way.
>>>
>>> Let us suppose that the renderer implemented a further category:
>>>
>>>    mode
>>>        mode-visual
>>>        mode-audio
>>>
>>> and applied the value "mode-audio" to all of the above signals, but
>>> only added the following signals with "mode-visual":
>>>
>>>    reason & duration-normal & locale & mode-visual
>>>    reason & duration-short & locale & mode-visual
>>>    reason & duration-long & locale & mode-visual
>>>
>>> That is, the only variation provided is in "duration".
>>>
>>> Then the indication "reason-auto.callback:duration-short:locale-al"
>>> would generate the same acceptable set as above, that is, there would
>>> be no "mode-visual" signals.  The indication
>>> "mode-visual:reason-auto.callback:duration-short:locale-al" would
>>> generate the acceptable set:
>>>
>>>    reason & duration-short & locale & mode-visual
>>>
>>> This is because the indication implies that matching "mode" is more
>>> important than matching "reason" and "duration".  Once the selection
>>> for "mode-visual" has been done, no further selection can be done on
>>> "reason", but "duration-short" can be acted upon.
>>>
>>> In practice, for use with a hearing-impaired user, the initial set of
>>> signals would be restricted to those with "mode-visual".  In this
>>> example, we have assumed that the user considers "mode-visual" and
>>> "mode-audio" to be equally acceptable.
>>>
>>> If we configured the renderer to only use "mode-visual" signals, then
>>> the indication "reason-auto.callback:duration-short:locale-al" would
>>> select only the signal
>>>
>>>    reason & duration-short & locale & mode-visual
>>>
>>> In this configuration, the indication
>>> "mode-audio:reason-auto.callback:duration-short:locale-al" would also
>>> select that signal, because the value "mode-audio" cannot be acted
>>> upon within that set of available signals.  It is effectively ignored.
>>>
>>> * Extensibility
>>>
>>> Let us now see how this scheme supports the various types of
>>> extensibility listed in Requirement C.
>>>
>>> - add further values to any existing category
>>>
>>> We could define an additional value such as
>>>
>>>    reason-avaya.special
>>>
>>> If the renderer had any signals that had this value, they would be
>>> selected by an indicator containing "reason-avaya.special".  If the
>>> renderer has no such signals, then in an indicator this value acts the
>>> same as "reason", that is, it has no effect.
>>>
>>> - add further categories that are orthogonal to existing categories
>>>
>>> We could define an additional category with values:
>>>
>>>    avaya
>>>        avaya-a
>>>        avaya-b
>>>        avaya-c
>>>
>>> If a renderer does not know of this category, all of its signals
>>> implicitly have the value "avaya".  Because of this, when such a
>>> renderer processes a value of this category in an indicator, it never
>>> changes the available set.
>>>
>>> - semantically subdivide the meaning provided by any existing
>>>  indication
>>>
>>> We could define subvalues of existing values, such as:
>>>
>>>    state
>>>        state-waiting
>>>            state-waiting-avaya.standard
>>>            state-waiting-avaya.vip
>>>
>>> If the renderer has two signals that differ only in the values
>>> "state-waiting-avaya.standard" and "state-waiting-avaya.vip", any
>>> indicator that does not know of this subdivision will never separate
>>> the two signals; they will both remain in the acceptable set or both
>>> be excluded from it.  So the renderer needs a policy for choosing one
>>> over the other -- In this case, clearly "state-waiting-avaya.standard"
>>> is to be preferred.
>>>
>>> However, if the indicator gives one of the two subvalues, the renderer
>>> will select the appropriate signal.
>>>
>>> If the indicator gives one of the two subvalues but the renderer only
>>> knows of "state-waiting", the "avaya.standard" or "avaya.vip" part of
>>> the value will never affect the acceptable set (since to select based
>>> on it would make the acceptable set empty).  Processing of
>>> "state-waiting-avaya.standard" would have the same effect as
>>> "state-waiting".
>>>
>>> - add further "value spaces" of indications whose semantics are not
>>>  related to existing indications, and thus, whose categories differ
>>>  from and do not interact with existing categories
>>>
>>> We can define completely independent spaces of signals by defining new
>>> categories to describe the new space, giving signals in one space the
>>> universal value for categories describing the other space.
>>>
>>> For example, we can define
>>>
>>>    auto
>>>        auto-make
>>>            auto-make-volvo
>>>            auto-make-toyota
>>>    color
>>>        color-red
>>>        color-green
>>>
>>> If we have signals with the values
>>>
>>>    auto-make-volvo & color-red
>>>    auto-make-volvo & color-green
>>>    auto-make-toyota & color-red
>>>    auto-make-toyota & color-green
>>>
>>> and add them to the set of "mode-visual" signals given above, we get:
>>>
>>>    reason & duration-normal & locale & mode-visual & auto & color
>>>    reason & duration-short & locale & mode-visual & auto & color
>>>    reason & duration-long & locale & mode-visual & auto & color
>>>    reason & duration & locale & mode & auto-make-volvo & color-red
>>>    reason & duration & locale & mode & auto-make-volvo & color-green
>>>    reason & duration & locale & mode & auto-make-toyota & color-red
>>>    reason & duration & locale & mode & auto-make-toyota & color-green
>>>
>>> This set is a combination of chalk and cheese.  But the first
>>> non-universal value in the indicator will select signals of one type
>>> or the other.  For instance, "auto-make" will select
>>>
>>>    reason & duration & locale & mode & auto-make-volvo & color-red
>>>    reason & duration & locale & mode & auto-make-volvo & color-green
>>>    reason & duration & locale & mode & auto-make-toyota & color-red
>>>    reason & duration & locale & mode & auto-make-toyota & color-green
>>>
>>> and "duration-normal" will select
>>>
>>>    reason & duration-normal & locale & mode-visual & auto & color
>>>
>>> Dale
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 

From john.elwell@siemens-enterprise.com  Wed Jun 16 06:44:48 2010
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To: Cullen Jennings <fluffy@cisco.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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When Cullen raised the question, I was wondering whether two dialogs are re=
lated if the media from the session associated with one dialog is forwarded=
 to the session associated with the other dialog. But then I realised that =
this would not apply if you wanted to correlate two dialogs that are not se=
ssion-related, e.g., two SUBSCRIBE-initiated dialogs. But on the other hand=
, the term session-ID hardly seems the right term for two related SUBSCRIBE=
-initiated dialogs, since there is no session involved.

So the first question to be asked is whether the work should really be limi=
ted to sessions, or whether it could apply to other types of dialog, or eve=
n to non-dialog-forming transactions such as MESSAGE and PUBLISH when they =
cross a B2BUA. Depending on the answer to this, the name and definition of =
the object we are trying to describe can be selected.

John


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: 15 June 2010 15:29
> To: Christer Holmberg
> Cc: DISPATCH list
> Subject: Re: [dispatch] Final Charter for Session-ID?
>=20
>=20
> I'm not following what you mean but it seems like this might=20
> lead to a workable definition. Can you propose something a=20
> bit more specific?
>=20
>=20
> On Jun 10, 2010, at 6:57 AM, Christer Holmberg wrote:
>=20
> >=20
> > Hi,
> >=20
> > Isn't the linkage the fact that C is the person A addressed=20
> for this specific session?
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org=20
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> >> Sent: 10. kes=E4kuuta 2010 15:54
> >> To: Muthu ArulMozhi Perumal (mperumal)
> >> Cc: DISPATCH list
> >> Subject: Re: [dispatch] Final Charter for Session-ID?
> >>=20
> >>=20
> >>=20
> >> Muthu ArulMozhi Perumal (mperumal) wrote:
> >>> Here is an initial attempt:
> >>>=20
> >>> If there is a dialog between A and B then A and B are in=20
> a session.
> >>> If there is a dialog between A and B and a dialog between B=20
> >> and C at=20
> >>> the same time then A, B and C are in a session.
> >>> More generally, if two or more participants are part of a=20
> session S=20
> >>> and at the same time there is a dialog between one of the=20
> >> participants=20
> >>> of S and another participant Z who is not part of S, then S=20
> >> (union) Z=20
> >>> is a session.
> >>=20
> >> The above just reiterates the problem that Cullen was raising.
> >> Just because B has a dialog with A and a dialog with C does=20
> >> not mean that there is a session between A and C. A gateway=20
> >> provides a perfect example of that.
> >>=20
> >> There needs to be some semantic link between the dialogs. The=20
> >> question is defining what sorts of semantic links should be=20
> >> used to share a session id.
> >>=20
> >> 	Thanks,
> >> 	Paul
> >>=20
> >>> I think this would cover most of SBC, 3PCC an app server=20
> >> cases. People=20
> >>> can add cases like park/retrieve if they think they should be=20
> >>> considered as part of a single session.
> >>>=20
> >>> Muthu
> >>>=20
> >>> |-----Original Message-----
> >>> |From: dispatch-bounces@ietf.org=20
> >> [mailto:dispatch-bounces@ietf.org] On
> >>> Behalf
> >>> |Of Cullen Jennings (fluffy)
> >>> |Sent: Wednesday, June 09, 2010 8:23 PM
> >>> |To: DISPATCH list
> >>> |Subject: Re: [dispatch] Final Charter for Session-ID?
> >>> |
> >>> |I have a problem with this proposed charter.
> >>> |
> >>> |On Jun 7, 2010, at 7:51 AM, Hadriel Kaplan wrote:
> >>> |
> >>> |>
> >>> |> The definition of a higher-layer "session" of the same=20
> >> "message-flow"
> >>> is
> >>> |not defined by SIP when it involves a B2BUA, and is difficult to
> >>> normatively
> >>> |define in practice.  For the purposes of this WG charter's=20
> >> scope, two
> >>> or
> >>> |more dialogs of a B2BUA are correlated under any=20
> conditions where=20
> >>> |correlation might aid troubleshooting, by identifying them as=20
> >>> |belonging
> >>> to
> >>> |the same higher-level session. Any solution documents from this=20
> >>> |working group may expand or constrain the scope of their=20
> >> mechanism's
> >>> correlation, if
> >>> |the working group so decides.
> >>> |
> >>> |I think the above text is way too vague and impossible to=20
> >> reduce to=20
> >>> |engineering practice. If you have 10 calls going to a PSTN=20
> >> GWs that=20
> >>> |is having voice quality problems, clearly having all of=20
> >> them grouped
> >>> together
> >>> |is useful in debugging. Yet I doubt we want all the calls from a=20
> >>> |given
> >>> PSTN
> >>> |GW to have the same correlation ID. Our lack of any solid=20
> >> definition=20
> >>> |of
> >>> what
> >>> |it is we are trying to correlate is a strong indication to=20
> >> me that we
> >>> have
> >>> |not yet come to any consensus about what the problem is=20
> >> this WG would
> >>> be
> >>> |trying to fix.
> >>> |
> >>> |I would like to see a WG charted in this area but I would=20
> >> expect the
> >>> IESG to
> >>> |more or less laugh at a problem statement that reduced=20
> down to "uh
> >>> might be
> >>> |useful for debuggin". Can some people please try and=20
> propose a more
> >>> concrete
> >>> |definition of when two sip dialogs will be correlated?
> >>> |
> >>> |Cullen
> >>> |
> >>> |_______________________________________________
> >>> |dispatch mailing list
> >>> |dispatch@ietf.org
> >>> |https://www.ietf.org/mailman/listinfo/dispatch
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>=20
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From laura.liess.dt@googlemail.com  Thu Jun 17 04:21:28 2010
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Date: Thu, 17 Jun 2010 13:15:17 +0200
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To: Paul Kyzivat <pkyzivat@cisco.com>, dispatch@ietf.org
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Dale,
Please see comments inline.

2010/6/16 WORLEY, Dale R (Dale) <dworley@avaya.com>:
>> From: Paul Kyzivat
>
>> IIUC, in what Dale has proposed the "indication" includes a prioritized
>> list of capability-values. If we followed the syntax of the current
>> draft, then the prioritization would probably come by considering the
>> order of URNs in the Alert-Info header to be significant.
>
> I would very much like to avoid having different URNs within the
> Alert-Info header interact; that seems to me to be against the concept
> of each URN being a distinct name. =A0(Which is why I am assuming that
> the syntax allows one URN to contain multiple "values".)

We had URN to contain multiple "values" in the initial proposal, see
http://tools.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt,
but only to some degree. We had there duration and priority in one
URN, but for combining call-waiting and duration we still needed two
URNs because they were different categories. For this version, we got
following comment from Adam:
http://www.ietf.org/mail-archive/web/bliss/current/msg01027.html . For
the next version of the draft, =A0Alan and I agreed to change to
combinations of discrete tokens.
So upto now we discussed two alternatives:
1) Return to URNs which contain multiple values, maybe with a modified
syntax to avoid multiple URNs. Problem: We have to find a way to avoid
the problem Adam pointed out to.
=A02) Or we can stay with the current model of different interacting
URNs within the Alert-Info header. Problem: We need to define proper
combination and rendering rules.

I think both could work fine if we can solve the corresponding problem.


>> One thing that *isn't* covered by Dale, or the above, is what to do if
>> the "renderer" disagrees with the "specifier" about the priorities. I
>> don't (yet) have an opinion if that is a problem that needs solving.
>
> Well, strictly speaking, the renderer doesn't have to follow the
> algorithm -- the normative part of the RFC will spell out what the
> intention of the URN is and what the constraints on renderers are, and
> we will put the algorithm in a non-normative appendix to prove that
> conforming renderers can be implemented straightforwardly.
>
>> From: Laura Liess
>
>> the problem I have with this is that we have
>> urn:alert:service:call-waiting already implemented and already
>> specified in 3GPP.
>
> If that is the *only* such URN defined and implemented in 3GPP, I
> expect that we can adjust the new syntax to give that one URN the
> intended interpretation while still implementing the semantics we
> agree upon within the whole namespace.
>
> Even if the syntax we decide upon is completely incompatible with this
> one URN, we can explicitly define it as a special case. =A0In the most
> extreme case, the RFC would define a *different* URN namespace, and we
> would specify that urn:alert:service:call-waiting maps to a specific
> URN in the new namespace.

I would prefer this to fit in the general scheme we are going to define.
>
> The harder situation is if the specifier wants to indicate some
> particular variant of the call-waiting signal. =A0What URN does it
> provide that include the variant information while still appearing to
> be "urn:alert:service:call-waiting" to 3GPP-only renderers?

I hope understand your question correctly.
Within 3GPP, there are no variants of the call-waiting signal. I am
not aware of any being used or needed by the DT.
This is probably a issue for PBXes, e.g. a short and =A0delayed
call-waiting signal. (Is this what you mean by "particular variant of
the call-waiting signal"?) =A0With both syntax alternatives we have up
to now, we would need at least two URNs. What you look for is getting
this all in only one URN, right? =A0Then we probably need a completely
new syntax. Maybe it's possible. We just have to be carefull to get
the problem Adam pointed out to solved.

Laura

>
> Dale
>

From pkyzivat@cisco.com  Thu Jun 17 06:44:31 2010
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Hmm. For some reason I have not received Dale's reply that Laura is 
replying to below. So I'm responding here.

	Thanks,
	Paul

Laura Liess wrote:
> Dale,
> Please see comments inline.
> 
> 2010/6/16 WORLEY, Dale R (Dale) <dworley@avaya.com>:
>>> From: Paul Kyzivat
>>> IIUC, in what Dale has proposed the "indication" includes a prioritized
>>> list of capability-values. If we followed the syntax of the current
>>> draft, then the prioritization would probably come by considering the
>>> order of URNs in the Alert-Info header to be significant.
>> I would very much like to avoid having different URNs within the
>> Alert-Info header interact; that seems to me to be against the concept
>> of each URN being a distinct name.  (Which is why I am assuming that
>> the syntax allows one URN to contain multiple "values".)

I don't see a problem there. In the general case, if there are multiple 
URIs then the recipient has to decide how to rationalize them into 
something to be done. So what is wrong with giving some guidance in that 
regard?

Requiring all the categories to be specified in one URN has the downside 
that all category values need to be standardized as part of a single URN 
spec. Conversely, consolidating multiple URNs gives the potential to 
define a new category via an entirely new URN scheme. (I guess we should 
discuss if that is a good thing or a bad thing.)

> We had URN to contain multiple "values" in the initial proposal, see
> http://tools.ietf.org/id/draft-alexeitsev-bliss-alert-info-urns-02.txt,
> but only to some degree. We had there duration and priority in one
> URN, but for combining call-waiting and duration we still needed two
> URNs because they were different categories. For this version, we got
> following comment from Adam:
> http://www.ietf.org/mail-archive/web/bliss/current/msg01027.html . For
> the next version of the draft,  Alan and I agreed to change to
> combinations of discrete tokens.
> So upto now we discussed two alternatives:
> 1) Return to URNs which contain multiple values, maybe with a modified
> syntax to avoid multiple URNs. Problem: We have to find a way to avoid
> the problem Adam pointed out to.
>  2) Or we can stay with the current model of different interacting
> URNs within the Alert-Info header. Problem: We need to define proper
> combination and rendering rules.
> 
> I think both could work fine if we can solve the corresponding problem.
> 
> 
>>> One thing that *isn't* covered by Dale, or the above, is what to do if
>>> the "renderer" disagrees with the "specifier" about the priorities. I
>>> don't (yet) have an opinion if that is a problem that needs solving.
>> Well, strictly speaking, the renderer doesn't have to follow the
>> algorithm -- the normative part of the RFC will spell out what the
>> intention of the URN is and what the constraints on renderers are, and
>> we will put the algorithm in a non-normative appendix to prove that
>> conforming renderers can be implemented straightforwardly.

OK. You have convinced me this isn't a problem. Good!

>>> From: Laura Liess
>>> the problem I have with this is that we have
>>> urn:alert:service:call-waiting already implemented and already
>>> specified in 3GPP.
>> If that is the *only* such URN defined and implemented in 3GPP, I
>> expect that we can adjust the new syntax to give that one URN the
>> intended interpretation while still implementing the semantics we
>> agree upon within the whole namespace.
>>
>> Even if the syntax we decide upon is completely incompatible with this
>> one URN, we can explicitly define it as a special case.  In the most
>> extreme case, the RFC would define a *different* URN namespace, and we
>> would specify that urn:alert:service:call-waiting maps to a specific
>> URN in the new namespace.
> 
> I would prefer this to fit in the general scheme we are going to define.
>> The harder situation is if the specifier wants to indicate some
>> particular variant of the call-waiting signal.  What URN does it
>> provide that include the variant information while still appearing to
>> be "urn:alert:service:call-waiting" to 3GPP-only renderers?
> 
> I hope understand your question correctly.
> Within 3GPP, there are no variants of the call-waiting signal. I am
> not aware of any being used or needed by the DT.
> This is probably a issue for PBXes, e.g. a short and  delayed
> call-waiting signal. (Is this what you mean by "particular variant of
> the call-waiting signal"?)  With both syntax alternatives we have up
> to now, we would need at least two URNs. What you look for is getting
> this all in only one URN, right?  Then we probably need a completely
> new syntax. Maybe it's possible. We just have to be carefull to get
> the problem Adam pointed out to solved.
> 
> Laura
> 
>> Dale
>>
> 

From john.elwell@siemens-enterprise.com  Thu Jun 17 09:24:28 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Cullen Jennings <fluffy@cisco.com>, DISPATCH list <dispatch@ietf.org>
Date: Thu, 17 Jun 2010 18:24:30 +0200
Thread-Topic: [dispatch] VIPR - proposed charter version 2
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Subject: Re: [dispatch] VIPR - proposed charter version 2
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"The VIPR WG will address this problem by developing a peer to peer based
approach to finding domains that claim to be responsible for a given
phone number..."
Does it necessarily have to be a peer-to-peer based approach? The ENUM/DNS =
approach is still a good approach for querying, but the issue is that with =
public ENUM the database simply doesn't get populated. So the key issue is =
finding a secure and attractive way of populating the database. Does the ch=
arter really need to include the words "peer to peer based", or should that=
 be left as a matter for discussion at the solution stage?

John
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: 15 June 2010 18:25
> To: DISPATCH list
> Subject: [dispatch] VIPR - proposed charter version 2
>=20
>=20
> Thanks for all the comments on the list. Here is a revised version:
>=20
> (PS. If anyone else wants to edit, this is in SVN and glad to=20
> get anyone write permission)
>=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> ViPR Charter Proposal (Version 2)
>=20
> WG Name:  Verification Involving PSTN Reachability (VIPR)
>=20
> There are two globally deployed address spaces for communications that
> more than a billion people use on a daily basis. They are=20
> phone numbers
> and DNS rooted address such as web servers and email addresses. The
> inter-domain signaling design of SIP is primarily designed for email
> style addresses yet a large percentage of SIP deployments mostly use
> phone numbers for identifying users. The goal of this working group is
> to enable inter-domains communications over the the internet, using
> protocol such as SIP, while still allowing people to use phone numbers
> to identify the person they wish to communicate with.
>=20
> The VIPR WG will address this problem by developing a peer to=20
> peer based
> approach to finding domains that claim to be responsible for a given
> phone number and validation protocols to ensure a reasonable=20
> likelihood
> that a given domain actually is responsible for the phone number. In
> this context, we use "responsible" to mean an administrative domain
> would would be at least one of the administrative domains that a PSTN
> call to this phone number would be routed to. Once the domain
> responsible for the phone number is found, existing protocols such as
> SIP and XMPP can be used for inter-domain communications.
>=20
> Some validation protocols may be based on knowledge gathered around a
> SIP call, for example, the ability to prove a call was=20
> received over the
> PSTN based on start and stop times. Other validation schemes, such as
> examining fingerprints or watermarking of PSTN media, to show that a
> domain received a particular PSTN phone call may also be considered by
> the working group. The WG will select and standardize at least one
> validation scheme.
>=20
> To help mitigate SPAM issues when using SIP between domains,=20
> the WG will
> define a mechanisms to enable one domain to check that incoming SIP
> messages from a different domain are coming from a domain that
> successfully validated a phone number. The working group will=20
> define new
> SIP headers and option tags to enable this.
>=20
> The essential characteristic of ViPR is establishing authentication by
> PSTN reachability when it is not possible to use a direct reference to
> ENUM databases or other direct assertions of PSTN number
> ownership. Elements such as public ENUM easily coexist with=20
> VIPR but no
> direct interaction with ENUM will be required.
>=20
> The problem statement and some possible starting points for solutions
> are further discussed in the following internet drafts which=20
> shall form
> the bases of the WG documents:
>=20
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
>=20
> The working group will carefully coordinate with the security=20
> area, O&M
> area, as well as the appropriate RAI WG including sipcore and p2psip.
>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Cullen Jennings <fluffy@cisco.com>, DISPATCH list <dispatch@ietf.org>
Date: Thu, 17 Jun 2010 14:59:02 -0400
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of El=
well, John [john.elwell@siemens-enterprise.com]

Does it necessarily have to be a peer-to-peer based approach?
_______________________________________________

I'd say the essence of VIPR is that it's peer-to-peer.  Or rather, that it =
enables the originator to succeed without the existence of a central or rel=
iable database.  (A peer-to-peer structure is used as a way for terminators=
 to advertise their claimed addresses.)  If we want to develop a database-c=
entered approach, it would be disjoint from the design effort for VIPR.

Dale

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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Laura Liess <laura.liess.dt@googlemail.com>
Date: Thu, 17 Jun 2010 15:32:22 -0400
Thread-Topic: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Pa=
ul Kyzivat [pkyzivat@cisco.com]

> I don't see a problem there. In the general case, if there are multiple
> URIs then the recipient has to decide how to rationalize them into
> something to be done. So what is wrong with giving some guidance in that
> regard?

We could do that, but it seems to me to be opening a can of worms, in
that the guidance would be (in the general case) how to rationalize an
arbitrary set of URIs.  If we limit the rationalization process to the
elements of a single URN of a particular namespace, the problem
becomes simpler.

> Requiring all the categories to be specified in one URN has the downside
> that all category values need to be standardized as part of a single URN
> spec.

Strictly speaking, you don't have to define the entire set of standard
categories in one RFC.  You do need some sort of registry for them and
their standardized values, but you can extend the registry with new
RFCs.  Of course, the vendor-specific extensions aren't cataloged
anywhere.

> Conversely, consolidating multiple URNs gives the potential to
> define a new category via an entirely new URN scheme. (I guess we
> should discuss if that is a good thing or a bad thing.)

I'd vote for considering that to be a bad thing.

Dale

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Subject: Re: [dispatch] VIPR - proposed charter version 2
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In general I think we should make charters  as specific as possible  but =
still broad enough to encompass the various types of solutions we want =
to consider for the problem. I think that helps things get done sooner.=20=


So if someone had a proposal on the table to solve this in some non peer =
to peer way, it would be easier to decide if we should do it in the same =
WG or something different. But we been talking about ideas for this for =
6 months now and no one has put together a proposal for some non peer to =
peer approach. I've heard lots of proposal to variations of the basic =
vipr scheme but they all leverages to peer to peer approach. I don't =
want a charter that is so vague that it is a research project. If =
someone has a specific approach that they think we should consider, then =
I'd certainly want to talk about that and adjust the proposed charter =
accordingly but, lacking that, I prefer a charter that makes it pretty =
clear what the WG is going to go do and not an exploratory type charter.=20=


As far as solving the problem with a centralized database, I think the =
DRINKS/SPEERMINT folks are pretty much doing the standardization works =
that would be needed for that approach. For the places that will work, =
it's an easier approach than VIPR which is looking for an approach for =
the places that does not make business sense.=20

Cullen

PS - I know what you mean by peer to peer here but don't even get me =
going about the difficulties of trying to define if something is peer to =
peer or not :-)=20


On Jun 17, 2010, at 10:24 AM, Elwell, John wrote:

> "The VIPR WG will address this problem by developing a peer to peer =
based
> approach to finding domains that claim to be responsible for a given
> phone number..."
> Does it necessarily have to be a peer-to-peer based approach? The =
ENUM/DNS approach is still a good approach for querying, but the issue =
is that with public ENUM the database simply doesn't get populated. So =
the key issue is finding a secure and attractive way of populating the =
database. Does the charter really need to include the words "peer to =
peer based", or should that be left as a matter for discussion at the =
solution stage?
>=20
> John
>=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Cullen Jennings
> > Sent: 15 June 2010 18:25
> > To: DISPATCH list
> > Subject: [dispatch] VIPR - proposed charter version 2
> >
> >
> > Thanks for all the comments on the list. Here is a revised version:
> >
> > (PS. If anyone else wants to edit, this is in SVN and glad to
> > get anyone write permission)
> >
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >
> > ViPR Charter Proposal (Version 2)
> >
> > WG Name:  Verification Involving PSTN Reachability (VIPR)
> >
> > There are two globally deployed address spaces for communications =
that
> > more than a billion people use on a daily basis. They are
> > phone numbers
> > and DNS rooted address such as web servers and email addresses. The
> > inter-domain signaling design of SIP is primarily designed for email
> > style addresses yet a large percentage of SIP deployments mostly use
> > phone numbers for identifying users. The goal of this working group =
is
> > to enable inter-domains communications over the the internet, using
> > protocol such as SIP, while still allowing people to use phone =
numbers
> > to identify the person they wish to communicate with.
> >
> > The VIPR WG will address this problem by developing a peer to
> > peer based
> > approach to finding domains that claim to be responsible for a given
> > phone number and validation protocols to ensure a reasonable
> > likelihood
> > that a given domain actually is responsible for the phone number. In
> > this context, we use "responsible" to mean an administrative domain
> > would would be at least one of the administrative domains that a =
PSTN
> > call to this phone number would be routed to. Once the domain
> > responsible for the phone number is found, existing protocols such =
as
> > SIP and XMPP can be used for inter-domain communications.
> >
> > Some validation protocols may be based on knowledge gathered around =
a
> > SIP call, for example, the ability to prove a call was
> > received over the
> > PSTN based on start and stop times. Other validation schemes, such =
as
> > examining fingerprints or watermarking of PSTN media, to show that a
> > domain received a particular PSTN phone call may also be considered =
by
> > the working group. The WG will select and standardize at least one
> > validation scheme.
> >
> > To help mitigate SPAM issues when using SIP between domains,
> > the WG will
> > define a mechanisms to enable one domain to check that incoming SIP
> > messages from a different domain are coming from a domain that
> > successfully validated a phone number. The working group will
> > define new
> > SIP headers and option tags to enable this.
> >
> > The essential characteristic of ViPR is establishing authentication =
by
> > PSTN reachability when it is not possible to use a direct reference =
to
> > ENUM databases or other direct assertions of PSTN number
> > ownership. Elements such as public ENUM easily coexist with
> > VIPR but no
> > direct interaction with ENUM will be required.
> >
> > The problem statement and some possible starting points for =
solutions
> > are further discussed in the following internet drafts which
> > shall form
> > the bases of the WG documents:
> >
> > draft-rosenberg-dispatch-vipr-overview
> > draft-rosenberg-dispatch-vipr-reload-usage
> > draft-rosenberg-dispatch-vipr-pvp
> > draft-rosenberg-dispatch-vipr-sip-antispam
> >
> > The working group will carefully coordinate with the security
> > area, O&M
> > area, as well as the appropriate RAI WG including sipcore and =
p2psip.
> >
> >
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
>=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From pkyzivat@cisco.com  Thu Jun 17 13:13:18 2010
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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I don't feel especially strongly one way or the other regarding the 
multiple URNs vs. single URN issue.

	Thanks,
	Paul

WORLEY, Dale R (Dale) wrote:
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat [pkyzivat@cisco.com]
> 
>> I don't see a problem there. In the general case, if there are multiple
>> URIs then the recipient has to decide how to rationalize them into
>> something to be done. So what is wrong with giving some guidance in that
>> regard?
> 
> We could do that, but it seems to me to be opening a can of worms, in
> that the guidance would be (in the general case) how to rationalize an
> arbitrary set of URIs.  If we limit the rationalization process to the
> elements of a single URN of a particular namespace, the problem
> becomes simpler.
> 
>> Requiring all the categories to be specified in one URN has the downside
>> that all category values need to be standardized as part of a single URN
>> spec.
> 
> Strictly speaking, you don't have to define the entire set of standard
> categories in one RFC.  You do need some sort of registry for them and
> their standardized values, but you can extend the registry with new
> RFCs.  Of course, the vendor-specific extensions aren't cataloged
> anywhere.
> 
>> Conversely, consolidating multiple URNs gives the potential to
>> define a new category via an entirely new URN scheme. (I guess we
>> should discuss if that is a good thing or a bad thing.)
> 
> I'd vote for considering that to be a bad thing.
> 
> Dale
> 

From fluffy@cisco.com  Thu Jun 17 13:54:42 2010
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To: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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I'm having a real hard time actually understand what you are objecting =
too. More  inline ...

On Jun 16, 2010, at 12:22 AM, Roni Even wrote:

> Hi,
> I read the charter and the listed drafts. I have no problem with the =
first
> two paragraph of the charter
>=20
> "There are two globally deployed address spaces for communications =
that more
> than a billion people use on a daily basis. They are phone numbers and =
DNS
> rooted address such as web servers and email addresses. The =
inter-domain
> signaling design of SIP is primarily designed for email style =
addresses yet
> a large percentage of SIP deployments mostly use phone numbers for
> identifying users. The goal of this working group is to enable =
inter-domains
> communications over the internet, using protocol such as SIP, while =
still
> allowing people to use phone numbers to identify the person they wish =
to
> communicate with.
>=20
> The VIPR WG will address this problem by developing a peer to peer =
based
> approach to finding domains that claim to be responsible for a given =
phone
> number and validation protocols to ensure a reasonable likelihood that =
a
> given domain actually is responsible for the phone number."
>=20
> I have a concern about using PSTN infrastructure for the reachability. =
My
> understanding so far was that SIP is trying to provide a new way for =
end to
> end communication that will replace the existing circuit switch
> infrastructure. This proposal says that the way to achieve end to end
> connectivity requires to have an end to end PSTN frastructure.

No this proposal says one of the way we can make PSTN address usable in =
inter domain SIP is this. Clearly this approach would no longer be valid =
when the PSTN was gone but by that point, presumably people would be =
using more internet style address in SIP. Keep in mind the only thing =
this work is trying to solve is mapping a PSTN address to a internet =
address. When the PSTN is gone, that problem goes with it. However the =
PSTN is far from gone - if I had to bet, I would guess that phone =
numbers were around longer than v6 addresses (and than email style =
address outlive them both). The issue is phone numbers are used by =
humans which makes them very hard to transition off of while things like =
v6 addresses are easier to transition off of as they are come and go =
with the technologies that use them. My point being all these addresses =
are around for a long time and the IETF works on standards to help =
transition and interwork between them. The Telegram service was emulated =
on top of other technologies long after it had been supplanted by fax - =
the PSTN will be similar.=20

>=20
> The third paragraph is talking about validation using PSTN calls. I =
think
> that we can look at validation of number ownership but should say that
> requiring PSTN calls to do it is not the recommended approach.
> This will
> allow managing of PSTN numbers not in the scope of PSTN =
infrastructure. Even
> the PSTN network is using external protocol like SS7 to route calls =
using
> databases for achieving reachability so why not say we can use similar
> infrastructure that will be IP based.

We did standardize and IP based database approach - it's called public =
ENUM. However, from a practical point of view the only people that could =
do the validations for ENUM are the carriers that have every business =
incentive to not do it. Regulators do not have the right incentives to =
regulate it into existence. VIPR on the other hand only requires =
carriers to route calls. Routing calls is something they do today, it is =
in their business interest so they will likely continue doing it, and =
the regulation around common carriage often require them to route calls. =
Both the existing regulations and economics favor that the carriers will =
do what is needed to enable a solution that leverages the PSTN for =
validation.=20

>=20
> I can understand this as a temporary solution but not as a standard
> developed by the IETF.

How temporary do you think the PSTN is? How many years?  Do you think we =
should not be doing standards for WG like MARTINI that only WG document =
limited itself to phone numbers? Should we not have standards about SIP =
/ PSTN interoperability? I don't hear you objecting to any of the other =
work we do hat is "temporary" in the sense it is only useful as long as =
the PSTN is around.

Seriously, I read this and just feel like you are grasping at straws for =
something to objet to since this proposal came from Cisco.=20


>=20
>=20
> Thanks
>=20
> Roni Even
>=20


From fluffy@cisco.com  Thu Jun 17 13:54:50 2010
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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On Jun 16, 2010, at 5:53 AM, Peter Musgrave wrote:

> Perhaps we should make it clear in the charter that the work requires
> that a node using ViPR have the ability to make and receive PSTN calls
> (or interface to something that does)?

That seems like a very pragmatic and good approach. I'm always a fan of =
charters having truth in advertising.

>=20
> I think general authentication without using PSTN reachability is a
> great idea=20

Well, I think it might be more accurate to say it is "great problem". We =
don't really have any ideas yet on how to solve it other than public =
ENUM which is less successful than one might have hoped.=20


From Even.roni@huawei.com  Thu Jun 17 15:47:37 2010
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From: Roni Even <Even.roni@huawei.com>
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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Hi Cullen,
I am sorry that you feel this way about me but I do not take this personally
since I am used to being blamed on lots of things.

Maybe I was not clear about my view.
Section 5.1 of the vipr overview provides the key properties that the
solution requires and I think it is a very good summary of the requirements.
I also think that dialing using E.164 numbers is better in many deployments.
(My background in H.323 has led me to it and BTW I believe that any such
solution can be applicable to H.323)

>From this point the vipr overview suggests a solution that is hinted in the
charter.

The question I have is if basing the validation of ownership of the PSTN
number can only be based on an actual PSTN call or does the charter allow
also other ways to validate the ownership of an E.164 number and can this
number be only the PSTN phone number or can it be other E.164 allocated
without an attached PSTN capability. 

The proposed charter as I read it limits the scope to the case where the
same number is used as the PSTN and SIP number. Did I understand correctly?



Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Thursday, June 17, 2010 11:55 PM
> To: Roni Even
> Cc: 'DISPATCH list'
> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
> 
> 
> I'm having a real hard time actually understand what you are objecting
> too. More  inline ...
> 
> On Jun 16, 2010, at 12:22 AM, Roni Even wrote:
> 
> > Hi,
> > I read the charter and the listed drafts. I have no problem with the
> first
> > two paragraph of the charter
> >
> > "There are two globally deployed address spaces for communications
> that more
> > than a billion people use on a daily basis. They are phone numbers
> and DNS
> > rooted address such as web servers and email addresses. The inter-
> domain
> > signaling design of SIP is primarily designed for email style
> addresses yet
> > a large percentage of SIP deployments mostly use phone numbers for
> > identifying users. The goal of this working group is to enable inter-
> domains
> > communications over the internet, using protocol such as SIP, while
> still
> > allowing people to use phone numbers to identify the person they wish
> to
> > communicate with.
> >
> > The VIPR WG will address this problem by developing a peer to peer
> based
> > approach to finding domains that claim to be responsible for a given
> phone
> > number and validation protocols to ensure a reasonable likelihood
> that a
> > given domain actually is responsible for the phone number."
> >
> > I have a concern about using PSTN infrastructure for the
> reachability. My
> > understanding so far was that SIP is trying to provide a new way for
> end to
> > end communication that will replace the existing circuit switch
> > infrastructure. This proposal says that the way to achieve end to end
> > connectivity requires to have an end to end PSTN frastructure.
> 
> No this proposal says one of the way we can make PSTN address usable in
> inter domain SIP is this. Clearly this approach would no longer be
> valid when the PSTN was gone but by that point, presumably people would
> be using more internet style address in SIP. Keep in mind the only
> thing this work is trying to solve is mapping a PSTN address to a
> internet address. When the PSTN is gone, that problem goes with it.
> However the PSTN is far from gone - if I had to bet, I would guess that
> phone numbers were around longer than v6 addresses (and than email
> style address outlive them both). The issue is phone numbers are used
> by humans which makes them very hard to transition off of while things
> like v6 addresses are easier to transition off of as they are come and
> go with the technologies that use them. My point being all these
> addresses are around for a long time and the IETF works on standards to
> help transition and interwork between them. The Telegram service was
> emulated on top of ot
>  her technologies long after it had been supplanted by fax - the PSTN
> will be similar.
> 
> >
> > The third paragraph is talking about validation using PSTN calls. I
> think
> > that we can look at validation of number ownership but should say
> that
> > requiring PSTN calls to do it is not the recommended approach.
> > This will
> > allow managing of PSTN numbers not in the scope of PSTN
> infrastructure. Even
> > the PSTN network is using external protocol like SS7 to route calls
> using
> > databases for achieving reachability so why not say we can use
> similar
> > infrastructure that will be IP based.
> 
> We did standardize and IP based database approach - it's called public
> ENUM. However, from a practical point of view the only people that
> could do the validations for ENUM are the carriers that have every
> business incentive to not do it. Regulators do not have the right
> incentives to regulate it into existence. VIPR on the other hand only
> requires carriers to route calls. Routing calls is something they do
> today, it is in their business interest so they will likely continue
> doing it, and the regulation around common carriage often require them
> to route calls. Both the existing regulations and economics favor that
> the carriers will do what is needed to enable a solution that leverages
> the PSTN for validation.
> 
> >
> > I can understand this as a temporary solution but not as a standard
> > developed by the IETF.
> 
> How temporary do you think the PSTN is? How many years?  Do you think
> we should not be doing standards for WG like MARTINI that only WG
> document limited itself to phone numbers? Should we not have standards
> about SIP / PSTN interoperability? I don't hear you objecting to any of
> the other work we do hat is "temporary" in the sense it is only useful
> as long as the PSTN is around.
> 
> Seriously, I read this and just feel like you are grasping at straws
> for something to objet to since this proposal came from Cisco.
> 
> 
> >
> >
> > Thanks
> >
> > Roni Even
> >
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From jdrosen@jdrosen.net  Thu Jun 17 18:08:49 2010
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Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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Roni,

The key requirement of the solution is that:

1. it be possible to validate the mapping of phone number to a domain 
which is "responsible" for that number, based on the definition Cullen 
has provided, and that

2. such validation be possible using only access to publicly available 
open interfaces to the PSTN, so that the validation can be performed by 
any domain wishing to participate.

At the moment, the only publicly open interface to the PSTN is the 
ability to make a call. If you are aware of other open and publicly 
available interfaces by which we can perform validation, I'd love to 
hear about them :)

Thanks,
Jonathan R.

Roni Even wrote:
> Hi Cullen,
> I am sorry that you feel this way about me but I do not take this personally
> since I am used to being blamed on lots of things.
> 
> Maybe I was not clear about my view.
> Section 5.1 of the vipr overview provides the key properties that the
> solution requires and I think it is a very good summary of the requirements.
> I also think that dialing using E.164 numbers is better in many deployments.
> (My background in H.323 has led me to it and BTW I believe that any such
> solution can be applicable to H.323)
> 
> From this point the vipr overview suggests a solution that is hinted in the
> charter.
> 
> The question I have is if basing the validation of ownership of the PSTN
> number can only be based on an actual PSTN call or does the charter allow
> also other ways to validate the ownership of an E.164 number and can this
> number be only the PSTN phone number or can it be other E.164 allocated
> without an attached PSTN capability. 
> 
> The proposed charter as I read it limits the scope to the case where the
> same number is used as the PSTN and SIP number. Did I understand correctly?
> 
> 
> 
> Roni Even
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Cullen Jennings
>> Sent: Thursday, June 17, 2010 11:55 PM
>> To: Roni Even
>> Cc: 'DISPATCH list'
>> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
>>
>>
>> I'm having a real hard time actually understand what you are objecting
>> too. More  inline ...
>>
>> On Jun 16, 2010, at 12:22 AM, Roni Even wrote:
>>
>>> Hi,
>>> I read the charter and the listed drafts. I have no problem with the
>> first
>>> two paragraph of the charter
>>>
>>> "There are two globally deployed address spaces for communications
>> that more
>>> than a billion people use on a daily basis. They are phone numbers
>> and DNS
>>> rooted address such as web servers and email addresses. The inter-
>> domain
>>> signaling design of SIP is primarily designed for email style
>> addresses yet
>>> a large percentage of SIP deployments mostly use phone numbers for
>>> identifying users. The goal of this working group is to enable inter-
>> domains
>>> communications over the internet, using protocol such as SIP, while
>> still
>>> allowing people to use phone numbers to identify the person they wish
>> to
>>> communicate with.
>>>
>>> The VIPR WG will address this problem by developing a peer to peer
>> based
>>> approach to finding domains that claim to be responsible for a given
>> phone
>>> number and validation protocols to ensure a reasonable likelihood
>> that a
>>> given domain actually is responsible for the phone number."
>>>
>>> I have a concern about using PSTN infrastructure for the
>> reachability. My
>>> understanding so far was that SIP is trying to provide a new way for
>> end to
>>> end communication that will replace the existing circuit switch
>>> infrastructure. This proposal says that the way to achieve end to end
>>> connectivity requires to have an end to end PSTN frastructure.
>> No this proposal says one of the way we can make PSTN address usable in
>> inter domain SIP is this. Clearly this approach would no longer be
>> valid when the PSTN was gone but by that point, presumably people would
>> be using more internet style address in SIP. Keep in mind the only
>> thing this work is trying to solve is mapping a PSTN address to a
>> internet address. When the PSTN is gone, that problem goes with it.
>> However the PSTN is far from gone - if I had to bet, I would guess that
>> phone numbers were around longer than v6 addresses (and than email
>> style address outlive them both). The issue is phone numbers are used
>> by humans which makes them very hard to transition off of while things
>> like v6 addresses are easier to transition off of as they are come and
>> go with the technologies that use them. My point being all these
>> addresses are around for a long time and the IETF works on standards to
>> help transition and interwork between them. The Telegram service was
>> emulated on top of ot
>>  her technologies long after it had been supplanted by fax - the PSTN
>> will be similar.
>>
>>> The third paragraph is talking about validation using PSTN calls. I
>> think
>>> that we can look at validation of number ownership but should say
>> that
>>> requiring PSTN calls to do it is not the recommended approach.
>>> This will
>>> allow managing of PSTN numbers not in the scope of PSTN
>> infrastructure. Even
>>> the PSTN network is using external protocol like SS7 to route calls
>> using
>>> databases for achieving reachability so why not say we can use
>> similar
>>> infrastructure that will be IP based.
>> We did standardize and IP based database approach - it's called public
>> ENUM. However, from a practical point of view the only people that
>> could do the validations for ENUM are the carriers that have every
>> business incentive to not do it. Regulators do not have the right
>> incentives to regulate it into existence. VIPR on the other hand only
>> requires carriers to route calls. Routing calls is something they do
>> today, it is in their business interest so they will likely continue
>> doing it, and the regulation around common carriage often require them
>> to route calls. Both the existing regulations and economics favor that
>> the carriers will do what is needed to enable a solution that leverages
>> the PSTN for validation.
>>
>>> I can understand this as a temporary solution but not as a standard
>>> developed by the IETF.
>> How temporary do you think the PSTN is? How many years?  Do you think
>> we should not be doing standards for WG like MARTINI that only WG
>> document limited itself to phone numbers? Should we not have standards
>> about SIP / PSTN interoperability? I don't hear you objecting to any of
>> the other work we do hat is "temporary" in the sense it is only useful
>> as long as the PSTN is around.
>>
>> Seriously, I read this and just feel like you are grasping at straws
>> for something to objet to since this proposal came from Cisco.
>>
>>
>>>
>>> Thanks
>>>
>>> Roni Even
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

-- 
Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
Chief Technology Strategist                    Mobile: +1 (732) 766-2496
Skype                                          SkypeIn: +1 (408) 465-0361
jdrosen@skype.net                              http://www.skype.com
jdrosen@jdrosen.net                            http://www.jdrosen.net



From jdrosen@jdrosen.net  Thu Jun 17 18:16:38 2010
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Subject: Re: [dispatch] VIPR - proposed charter version 2
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Cullen,

This looks good. My major comment is that I do think we should include 
VAP. There are applications for VAP which are interdomain. Furthermore, 
even for the intra-domain case, this will be a common protocol for 
communicating between products made from disparate vendors. For example, 
PBX from vendor A talks to the ViPR server from vendor B.

Other minor comments:

Cullen Jennings wrote:

> style addresses yet a large percentage of SIP deployments mostly use
> phone numbers for identifying users. The goal of this working group is
> to enable inter-domains communications over the the internet, using

s/inter-domains/inter-domain

> protocol such as SIP, while still allowing people to use phone numbers

s/protocol/protocols

> to identify the person they wish to communicate with.
> 
> The VIPR WG will address this problem by developing a peer to peer based
> approach to finding domains that claim to be responsible for a given
> phone number and validation protocols to ensure a reasonable likelihood
> that a given domain actually is responsible for the phone number. In
> this context, we use "responsible" to mean an administrative domain
> would would be at least one of the administrative domains that a PSTN

double woulds.

> call to this phone number would be routed to. Once the domain
> responsible for the phone number is found, existing protocols such as
> SIP and XMPP can be used for inter-domain communications.

XMPP? really? does it support phone number addressing?

> 
> Some validation protocols may be based on knowledge gathered around a
> SIP call, for example, the ability to prove a call was received over the

I think you mean PSTN call, and not a SIP call.

> PSTN based on start and stop times. Other validation schemes, such as
> examining fingerprints or watermarking of PSTN media, to show that a
> domain received a particular PSTN phone call may also be considered by
> the working group. The WG will select and standardize at least one
> validation scheme.
> 
> To help mitigate SPAM issues when using SIP between domains, the WG will
> define a mechanisms to enable one domain to check that incoming SIP

s/a mechanisms/a mechanism

> messages from a different domain are coming from a domain that
> successfully validated a phone number. The working group will define new
> SIP headers and option tags to enable this.

The key characteristic is not actually that validation had occurred, but 
rather, that the calling domain had previously placed a PSTN call to the 
number in question.

> 
> The essential characteristic of ViPR is establishing authentication by
> PSTN reachability when it is not possible to use a direct reference to
> ENUM databases or other direct assertions of PSTN number
> ownership. Elements such as public ENUM easily coexist with VIPR but no
> direct interaction with ENUM will be required.

I'd like to get a clear statement in here about the fact that ViPR is 
incrementally deployable using only existing interfaces to the PSTN. No 
change to PSTN capabilities, or new access to its databases, or support 
from PSTN providers, is required in any way.

> 
> The problem statement and some possible starting points for solutions
> are further discussed in the following internet drafts which shall form
> the bases of the WG documents:
> 
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam

Add VAP.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
Chief Technology Strategist                    Mobile: +1 (732) 766-2496
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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On 6/17/10 7:44 AM, Paul Kyzivat wrote:

> Requiring all the categories to be specified in one URN has the downsid=
e
> that all category values need to be standardized as part of a single UR=
N
> spec. Conversely, consolidating multiple URNs gives the potential to
> define a new category via an entirely new URN scheme.=20

To clarify, there is no such thing as a "URN scheme". Upon approval,
draft-liess-dispatch-alert-info-urns will register a URN namespace
identifier (NID) in accordance with RFC 3406, not a "URN scheme".

Peter

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From: "Milinski, Alexander (NSN - DE/Munich)" <alexander.milinski@nsn.com>
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Cullen, Jonathan,

how is the PSTN defined in the sense of the charter?

Does it include the SIP networks existing today, which are
interconnected to the PSTN in the common sense (S=3D "switched"...) and
you can reach by dialling a phone number?

Regards,
Alex

=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Jonathan Rosenberg
> Sent: Friday, June 18, 2010 3:09 AM
> To: Roni Even
> Cc: 'DISPATCH list'
> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
>=20
> Roni,
>=20
> The key requirement of the solution is that:
>=20
> 1. it be possible to validate the mapping of phone number to a domain=20
> which is "responsible" for that number, based on the=20
> definition Cullen=20
> has provided, and that
>=20
> 2. such validation be possible using only access to publicly=20
> available=20
> open interfaces to the PSTN, so that the validation can be=20
> performed by=20
> any domain wishing to participate.
>=20
> At the moment, the only publicly open interface to the PSTN is the=20
> ability to make a call. If you are aware of other open and publicly=20
> available interfaces by which we can perform validation, I'd love to=20
> hear about them :)
>=20
> Thanks,
> Jonathan R.
>=20
> Roni Even wrote:
> > Hi Cullen,
> > I am sorry that you feel this way about me but I do not=20
> take this personally
> > since I am used to being blamed on lots of things.
> >=20
> > Maybe I was not clear about my view.
> > Section 5.1 of the vipr overview provides the key=20
> properties that the
> > solution requires and I think it is a very good summary of=20
> the requirements.
> > I also think that dialing using E.164 numbers is better in=20
> many deployments.
> > (My background in H.323 has led me to it and BTW I believe=20
> that any such
> > solution can be applicable to H.323)
> >=20
> > From this point the vipr overview suggests a solution that=20
> is hinted in the
> > charter.
> >=20
> > The question I have is if basing the validation of=20
> ownership of the PSTN
> > number can only be based on an actual PSTN call or does the=20
> charter allow
> > also other ways to validate the ownership of an E.164=20
> number and can this
> > number be only the PSTN phone number or can it be other=20
> E.164 allocated
> > without an attached PSTN capability.=20
> >=20
> > The proposed charter as I read it limits the scope to the=20
> case where the
> > same number is used as the PSTN and SIP number. Did I=20
> understand correctly?
> >=20
> >=20
> >=20
> > Roni Even
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On
> >> Behalf Of Cullen Jennings
> >> Sent: Thursday, June 17, 2010 11:55 PM
> >> To: Roni Even
> >> Cc: 'DISPATCH list'
> >> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
> >>
> >>
> >> I'm having a real hard time actually understand what you=20
> are objecting
> >> too. More  inline ...
> >>
> >> On Jun 16, 2010, at 12:22 AM, Roni Even wrote:
> >>
> >>> Hi,
> >>> I read the charter and the listed drafts. I have no=20
> problem with the
> >> first
> >>> two paragraph of the charter
> >>>
> >>> "There are two globally deployed address spaces for communications
> >> that more
> >>> than a billion people use on a daily basis. They are phone numbers
> >> and DNS
> >>> rooted address such as web servers and email addresses. The inter-
> >> domain
> >>> signaling design of SIP is primarily designed for email style
> >> addresses yet
> >>> a large percentage of SIP deployments mostly use phone numbers for
> >>> identifying users. The goal of this working group is to=20
> enable inter-
> >> domains
> >>> communications over the internet, using protocol such as=20
> SIP, while
> >> still
> >>> allowing people to use phone numbers to identify the=20
> person they wish
> >> to
> >>> communicate with.
> >>>
> >>> The VIPR WG will address this problem by developing a peer to peer
> >> based
> >>> approach to finding domains that claim to be responsible=20
> for a given
> >> phone
> >>> number and validation protocols to ensure a reasonable likelihood
> >> that a
> >>> given domain actually is responsible for the phone number."
> >>>
> >>> I have a concern about using PSTN infrastructure for the
> >> reachability. My
> >>> understanding so far was that SIP is trying to provide a=20
> new way for
> >> end to
> >>> end communication that will replace the existing circuit switch
> >>> infrastructure. This proposal says that the way to=20
> achieve end to end
> >>> connectivity requires to have an end to end PSTN frastructure.
> >> No this proposal says one of the way we can make PSTN=20
> address usable in
> >> inter domain SIP is this. Clearly this approach would no longer be
> >> valid when the PSTN was gone but by that point, presumably=20
> people would
> >> be using more internet style address in SIP. Keep in mind the only
> >> thing this work is trying to solve is mapping a PSTN address to a
> >> internet address. When the PSTN is gone, that problem goes with it.
> >> However the PSTN is far from gone - if I had to bet, I=20
> would guess that
> >> phone numbers were around longer than v6 addresses (and than email
> >> style address outlive them both). The issue is phone=20
> numbers are used
> >> by humans which makes them very hard to transition off of=20
> while things
> >> like v6 addresses are easier to transition off of as they=20
> are come and
> >> go with the technologies that use them. My point being all these
> >> addresses are around for a long time and the IETF works on=20
> standards to
> >> help transition and interwork between them. The Telegram=20
> service was
> >> emulated on top of ot
> >>  her technologies long after it had been supplanted by fax=20
> - the PSTN
> >> will be similar.
> >>
> >>> The third paragraph is talking about validation using=20
> PSTN calls. I
> >> think
> >>> that we can look at validation of number ownership but should say
> >> that
> >>> requiring PSTN calls to do it is not the recommended approach.
> >>> This will
> >>> allow managing of PSTN numbers not in the scope of PSTN
> >> infrastructure. Even
> >>> the PSTN network is using external protocol like SS7 to=20
> route calls
> >> using
> >>> databases for achieving reachability so why not say we can use
> >> similar
> >>> infrastructure that will be IP based.
> >> We did standardize and IP based database approach - it's=20
> called public
> >> ENUM. However, from a practical point of view the only people that
> >> could do the validations for ENUM are the carriers that have every
> >> business incentive to not do it. Regulators do not have the right
> >> incentives to regulate it into existence. VIPR on the=20
> other hand only
> >> requires carriers to route calls. Routing calls is=20
> something they do
> >> today, it is in their business interest so they will=20
> likely continue
> >> doing it, and the regulation around common carriage often=20
> require them
> >> to route calls. Both the existing regulations and=20
> economics favor that
> >> the carriers will do what is needed to enable a solution=20
> that leverages
> >> the PSTN for validation.
> >>
> >>> I can understand this as a temporary solution but not as=20
> a standard
> >>> developed by the IETF.
> >> How temporary do you think the PSTN is? How many years? =20
> Do you think
> >> we should not be doing standards for WG like MARTINI that only WG
> >> document limited itself to phone numbers? Should we not=20
> have standards
> >> about SIP / PSTN interoperability? I don't hear you=20
> objecting to any of
> >> the other work we do hat is "temporary" in the sense it is=20
> only useful
> >> as long as the PSTN is around.
> >>
> >> Seriously, I read this and just feel like you are grasping=20
> at straws
> >> for something to objet to since this proposal came from Cisco.
> >>
> >>
> >>>
> >>> Thanks
> >>>
> >>> Roni Even
> >>>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
> Chief Technology Strategist                    Mobile: +1=20
> (732) 766-2496
> Skype                                          SkypeIn: +1=20
> (408) 465-0361
> jdrosen@skype.net                              http://www.skype.com
> jdrosen@jdrosen.net                            http://www.jdrosen.net
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From pkyzivat@cisco.com  Fri Jun 18 06:15:35 2010
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Peter,

Ok - I couldn't remember the precise term and used something convenient.

	Thanks,
	Paul

Peter Saint-Andre wrote:
> On 6/17/10 7:44 AM, Paul Kyzivat wrote:
> 
>> Requiring all the categories to be specified in one URN has the downside
>> that all category values need to be standardized as part of a single URN
>> spec. Conversely, consolidating multiple URNs gives the potential to
>> define a new category via an entirely new URN scheme. 
> 
> To clarify, there is no such thing as a "URN scheme". Upon approval,
> draft-liess-dispatch-alert-info-urns will register a URN namespace
> identifier (NID) in accordance with RFC 3406, not a "URN scheme".
> 
> Peter
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From jdrosen@jdrosen.net  Fri Jun 18 06:52:47 2010
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Cc: DISPATCH list <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
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PSTN refers to the network which is the authoritative source of routing 
for calls made to E.164 numbers. Whether it uses SIP internally, or 
whether it provides SIP interfaces (aka SIP trunking), changes nothing.

-Jonathan R.

Milinski, Alexander (NSN - DE/Munich) wrote:
> Cullen, Jonathan,
> 
> how is the PSTN defined in the sense of the charter?
> 
> Does it include the SIP networks existing today, which are
> interconnected to the PSTN in the common sense (S= "switched"...) and
> you can reach by dialling a phone number?
> 
> Regards,
> Alex
> 
>  
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Jonathan Rosenberg
>> Sent: Friday, June 18, 2010 3:09 AM
>> To: Roni Even
>> Cc: 'DISPATCH list'
>> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
>>
>> Roni,
>>
>> The key requirement of the solution is that:
>>
>> 1. it be possible to validate the mapping of phone number to a domain 
>> which is "responsible" for that number, based on the 
>> definition Cullen 
>> has provided, and that
>>
>> 2. such validation be possible using only access to publicly 
>> available 
>> open interfaces to the PSTN, so that the validation can be 
>> performed by 
>> any domain wishing to participate.
>>
>> At the moment, the only publicly open interface to the PSTN is the 
>> ability to make a call. If you are aware of other open and publicly 
>> available interfaces by which we can perform validation, I'd love to 
>> hear about them :)
>>
>> Thanks,
>> Jonathan R.
>>
>> Roni Even wrote:
>>> Hi Cullen,
>>> I am sorry that you feel this way about me but I do not 
>> take this personally
>>> since I am used to being blamed on lots of things.
>>>
>>> Maybe I was not clear about my view.
>>> Section 5.1 of the vipr overview provides the key 
>> properties that the
>>> solution requires and I think it is a very good summary of 
>> the requirements.
>>> I also think that dialing using E.164 numbers is better in 
>> many deployments.
>>> (My background in H.323 has led me to it and BTW I believe 
>> that any such
>>> solution can be applicable to H.323)
>>>
>>> From this point the vipr overview suggests a solution that 
>> is hinted in the
>>> charter.
>>>
>>> The question I have is if basing the validation of 
>> ownership of the PSTN
>>> number can only be based on an actual PSTN call or does the 
>> charter allow
>>> also other ways to validate the ownership of an E.164 
>> number and can this
>>> number be only the PSTN phone number or can it be other 
>> E.164 allocated
>>> without an attached PSTN capability. 
>>>
>>> The proposed charter as I read it limits the scope to the 
>> case where the
>>> same number is used as the PSTN and SIP number. Did I 
>> understand correctly?
>>>
>>>
>>> Roni Even
>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On
>>>> Behalf Of Cullen Jennings
>>>> Sent: Thursday, June 17, 2010 11:55 PM
>>>> To: Roni Even
>>>> Cc: 'DISPATCH list'
>>>> Subject: Re: [dispatch] VIPR - proposed charter version 2 - PSTN?
>>>>
>>>>
>>>> I'm having a real hard time actually understand what you 
>> are objecting
>>>> too. More  inline ...
>>>>
>>>> On Jun 16, 2010, at 12:22 AM, Roni Even wrote:
>>>>
>>>>> Hi,
>>>>> I read the charter and the listed drafts. I have no 
>> problem with the
>>>> first
>>>>> two paragraph of the charter
>>>>>
>>>>> "There are two globally deployed address spaces for communications
>>>> that more
>>>>> than a billion people use on a daily basis. They are phone numbers
>>>> and DNS
>>>>> rooted address such as web servers and email addresses. The inter-
>>>> domain
>>>>> signaling design of SIP is primarily designed for email style
>>>> addresses yet
>>>>> a large percentage of SIP deployments mostly use phone numbers for
>>>>> identifying users. The goal of this working group is to 
>> enable inter-
>>>> domains
>>>>> communications over the internet, using protocol such as 
>> SIP, while
>>>> still
>>>>> allowing people to use phone numbers to identify the 
>> person they wish
>>>> to
>>>>> communicate with.
>>>>>
>>>>> The VIPR WG will address this problem by developing a peer to peer
>>>> based
>>>>> approach to finding domains that claim to be responsible 
>> for a given
>>>> phone
>>>>> number and validation protocols to ensure a reasonable likelihood
>>>> that a
>>>>> given domain actually is responsible for the phone number."
>>>>>
>>>>> I have a concern about using PSTN infrastructure for the
>>>> reachability. My
>>>>> understanding so far was that SIP is trying to provide a 
>> new way for
>>>> end to
>>>>> end communication that will replace the existing circuit switch
>>>>> infrastructure. This proposal says that the way to 
>> achieve end to end
>>>>> connectivity requires to have an end to end PSTN frastructure.
>>>> No this proposal says one of the way we can make PSTN 
>> address usable in
>>>> inter domain SIP is this. Clearly this approach would no longer be
>>>> valid when the PSTN was gone but by that point, presumably 
>> people would
>>>> be using more internet style address in SIP. Keep in mind the only
>>>> thing this work is trying to solve is mapping a PSTN address to a
>>>> internet address. When the PSTN is gone, that problem goes with it.
>>>> However the PSTN is far from gone - if I had to bet, I 
>> would guess that
>>>> phone numbers were around longer than v6 addresses (and than email
>>>> style address outlive them both). The issue is phone 
>> numbers are used
>>>> by humans which makes them very hard to transition off of 
>> while things
>>>> like v6 addresses are easier to transition off of as they 
>> are come and
>>>> go with the technologies that use them. My point being all these
>>>> addresses are around for a long time and the IETF works on 
>> standards to
>>>> help transition and interwork between them. The Telegram 
>> service was
>>>> emulated on top of ot
>>>>  her technologies long after it had been supplanted by fax 
>> - the PSTN
>>>> will be similar.
>>>>
>>>>> The third paragraph is talking about validation using 
>> PSTN calls. I
>>>> think
>>>>> that we can look at validation of number ownership but should say
>>>> that
>>>>> requiring PSTN calls to do it is not the recommended approach.
>>>>> This will
>>>>> allow managing of PSTN numbers not in the scope of PSTN
>>>> infrastructure. Even
>>>>> the PSTN network is using external protocol like SS7 to 
>> route calls
>>>> using
>>>>> databases for achieving reachability so why not say we can use
>>>> similar
>>>>> infrastructure that will be IP based.
>>>> We did standardize and IP based database approach - it's 
>> called public
>>>> ENUM. However, from a practical point of view the only people that
>>>> could do the validations for ENUM are the carriers that have every
>>>> business incentive to not do it. Regulators do not have the right
>>>> incentives to regulate it into existence. VIPR on the 
>> other hand only
>>>> requires carriers to route calls. Routing calls is 
>> something they do
>>>> today, it is in their business interest so they will 
>> likely continue
>>>> doing it, and the regulation around common carriage often 
>> require them
>>>> to route calls. Both the existing regulations and 
>> economics favor that
>>>> the carriers will do what is needed to enable a solution 
>> that leverages
>>>> the PSTN for validation.
>>>>
>>>>> I can understand this as a temporary solution but not as 
>> a standard
>>>>> developed by the IETF.
>>>> How temporary do you think the PSTN is? How many years?  
>> Do you think
>>>> we should not be doing standards for WG like MARTINI that only WG
>>>> document limited itself to phone numbers? Should we not 
>> have standards
>>>> about SIP / PSTN interoperability? I don't hear you 
>> objecting to any of
>>>> the other work we do hat is "temporary" in the sense it is 
>> only useful
>>>> as long as the PSTN is around.
>>>>
>>>> Seriously, I read this and just feel like you are grasping 
>> at straws
>>>> for something to objet to since this proposal came from Cisco.
>>>>
>>>>
>>>>> Thanks
>>>>>
>>>>> Roni Even
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
>> Chief Technology Strategist                    Mobile: +1 
>> (732) 766-2496
>> Skype                                          SkypeIn: +1 
>> (408) 465-0361
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From stpeter@stpeter.im  Fri Jun 18 07:07:11 2010
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Subject: Re: [dispatch] Conceptual proposal for extensibility of Alert-Info URNs
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Understood. The term "branch of the URN tree" might be an informal way
of describing a URN namespace identifier. But the charter proposal says
"URN scheme", too, so I just wanted to nip that error in the bud. :)

On 6/18/10 7:15 AM, Paul Kyzivat wrote:
> Peter,
>=20
> Ok - I couldn't remember the precise term and used something convenient=
=2E
>=20
>     Thanks,
>     Paul
>=20
> Peter Saint-Andre wrote:
>> On 6/17/10 7:44 AM, Paul Kyzivat wrote:
>>
>>> Requiring all the categories to be specified in one URN has the downs=
ide
>>> that all category values need to be standardized as part of a single =
URN
>>> spec. Conversely, consolidating multiple URNs gives the potential to
>>> define a new category via an entirely new URN scheme.=20
>>
>> To clarify, there is no such thing as a "URN scheme". Upon approval,
>> draft-liess-dispatch-alert-info-urns will register a URN namespace
>> identifier (NID) in accordance with RFC 3406, not a "URN scheme".
>>
>> Peter
>>
>>


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From gonzalo.camarillo@ericsson.com  Mon Jun 21 04:36:54 2010
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Date: Mon, 21 Jun 2010 14:36:20 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [dispatch] Status of WGs proposed to be chartered
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Folks,

FYI: the IESG has just approved the charter proposals for disaggregated
media, alert-info URNs, and user-to-user information for external
review. I have already requested the start of the external review
processes, which will start shortly. If you want to follow them, keep an
eye on the IETF announce and IETF discussion mailing lists.

Cheers,

Gonzalo


From mary.ietf.barnes@gmail.com  Mon Jun 21 14:59:53 2010
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Subject: [dispatch] Fwd: Nomcom 2010-2011: Second Call for Volunteers
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Hi all,

For folks that are not on the IETF announcement list, please consider
volunteering for the 2010-2011 Nomcom. This is your opportunity to
contribute to the community by participating in the selecting of the
next set of leaders (IESG, IAB, IAOC and IETF chair).

Thanks,
Mary
(as WG co-chair and Nomcom advisor for 2010-2011)


---------- Forwarded message ----------
From: NomCom Chair <nomcom-chair@ietf.org>
Date: Mon, Jun 21, 2010 at 4:49 PM
Subject: Nomcom 2010-2011: Second Call for Volunteers
To: IETF Announcement list <ietf-announce@ietf.org>


Hi all,

This is the Second call for Volunteers for the 2010-11 nomcom. =A0We are
just about halfway through the volunteer period so if you are considering
volunteering please do so very soon.

I am pleased to report that we have 29 volunteers thus far whose
qualifications have been confirmed by the secretariat. I have notified
each of these volunteers by email.

If you volunteered before 08:00 PDT (15:00 UTC) on June 21 to serve as a
voting member and have not received a confirmation email from me, please
re-submit and bring to my attention right away!

However, we need many more volunteers. =A0You have until 17:00 PDT (24:00
UTC) on July 8, 2010 to volunteer for nomcom but it is much better if you
volunteer as early as possible. =A0The more volunteers, the better chance w=
e
have of choosing a random yet representative cross section of the IETF
pouplation.

As a reminder, volunteers must have attended 3 of the past 5 IETF
meetings - per RFC 3777, which means 3 of the following meetings: IETF-73,
IETF-74, IETF-75, IETF-76 and IETF-77. =A0If you qualify, and are willing t=
o
forgo appointment to any of the positions for which the nominating
committee is responsible, please volunteer.

The 10 nominating committee members are selected randomly from a pool of
volunteers. =A0The details of the operation of nomcom may be found in RFC
3777. =A0The process on how to volunteer for nomcom is in the initial
announcement: =A0https://datatracker.ietf.org/ann/nomcom/2330/

The lists of open positions and people whose terms end in March 2011 and
thus the positions for which the nominating committee is responsible are
summarized in the initial announcement:
https://datatracker.ietf.org/ann/nomcom/2330/

The 29 volunteers who have thus far been qualified by the secretariat
are:

Fred Baker, Cisco Systems

Stephan Wenger, Vidyo, Inc.

Marc Blanchet, Viagenie

Lixia Zhang, UCLA

John Drake, Juniper Networks

Ole Troan, Cisco

Jiankang Yao, CNNIC

Wassim Haddad, Ericsson

Yi Zhao, Huawei USA

Teemu Savolainen, Nokia

Jouni Korhonen, Nokia Siemens Networks

Mehmet Ersue, Nokia Siemens Networks

Christian Schmidt, Nokia Siemens Networks

Stephen Hanna, Juniper Networks

Suresh Krishnan, Ericsson

Eric Gray, Ericsson

David Sinicrope, Ericsson

Jan Melen, Ericsson

Richard Barnes, BBN Technologies

Hugo Salgado Hernandez, NIC Chile

Ning Zong, Huawei Technologies

Qin Wu, Huawei Technologies

Karen Seo, BBN Technologies

Haibin Song, Huawei Technologies

Susan Hares, Huawei Technologies

Tony Hansen. AT&T Labs

Fuqing Huang, Huawei Technologies Co., Ltd.

Huub van Helvoort, Huawei Technologies

Miya Kohno, Juniper Networks

The primary activity for this nomcom will begin just prior to IETF-78 in
Maastricht and should be completed in early January 2011. =A0The nomcom wil=
l
be collecting requirements from the community, as well as talking to
candidates and to community members about candidates. There will be weekly
conference calls to ensure progress. Thus, being a nomcom member does
require some time commitment.

While, there is no requirement in RFC 3777 that a participant attend IETF
meetings while serving on nomcom, please consider that during the IETF
meetings, people that do not attend would be expected to remotely
participate during the day in the time zones of the meeting locations -
Maastricht at the end of July and Beijing in November.

If you are not yet sure you would like to volunteer, please consider that
nomcom members play a very important role in shaping the leadership of the
IETF. =A0Ensuring the leadership of the IETF is fair and balanced and
comprised of those who can lead the IETF in the right direction is an
important responsibility that rests on the IETF participants at large.
Volunteering for the nomcom is a good way of contributing in that
direction.

Please volunteer by sending an email before:
17:00 PDT (24:00 UTC) July 8, 2010 as follows:

To: nomcom-chair@ietf.org
Subject: Nomcom 2010-11 Volunteer

Please include the following information in the body:

<Your Full Name> =A0// As you enter in the IETF Registration Form,
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0// First/Given name followed by Last/Fam=
ily Name

<Current Primary Affiliation>
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0// typically what goes in the Company fi=
eld
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0// in the IETF Registration Form

[<all email addresses used to Register for the past 5 IETF meetings>]
<Preferred email address> =A0//

<Telephone number> =A0 =A0 =A0 =A0 // For confirmation if selected

Please expect an email response from me within 3 days stating whether or
not you are qualified. =A0If you don't receive a response, please re-send
your email with the tag "Duplicate:" added to the subject line.

Thank you and I hope to hear from you,

Thomas Walsh
Chair, Nomcom 2010-11
Email: nomcom-chair@ietf.org


_______________________________________________
IETF-Announce mailing list
IETF-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/ietf-announce

From henry.sinnreich@gmail.com  Tue Jun 22 08:30:34 2010
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Date: Tue, 22 Jun 2010 10:29:56 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
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Thread-Topic: SIP APIs for Communications on the Web
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Subject: [dispatch] SIP APIs for Communications on the Web
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

--B_3360047400_16103206
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	charset="ISO-8859-1"
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Following the discussions on the Dispatch list in February and also in
private, we have posted a new version of the I-D:

http://www.ietf.org/id/draft-sinnreich-sip-web-apis-01.txt

The clarifications include such topics as (in no particular order)
* SIP and/or (vs.) XMPP/Jingle
* HTML5 support for application scripts
* Bidirectional HTTP, IETF "HyBi" WG
* Hiding SIP complexity under the right, standards based API
* Drawing the line between basic usage scenarios for a simple API and yet
how to support complex VoIP call flows
* The SIP state machine moves to the business logic in UA and Web feature
servers; which is preferable
* The traffic and time spent with web applications far surpasses the time t=
o
talk on the phone (exceptions)
* Added references for metadata standards to replace SDP
* SDPng was already proposed in the early =8C90s
* Keep only UDP for media transport, RTP data about the media is moved to
the application as well
* New reference for HIP specific NAT traversal utilities.

Please let us know how you feel these and other issues have been addressed.

Abstract

   This memo describes a standards based approach to enable web based
   interactive multimedia communications.  The objective is giving web
   developers the software tools to add communication widgets to web
   pages.

   The proposed SIP API does not necessarily require SIP protocol
   expertise by web developers for basic multimedia communications
   though it can be extended to port complex VoIP services to the web.
   The SIP API can also support the transition from network
   infrastructure based VoIP to rich web based communications.

   The benefits of the formal REST architecture of the web are extended
   to real time communications.

   Only two standard application layer protocols are required: HTTP for
   signaling data communications such as in SIP and/or XMPP and UDP for
   real time media transport. We consider replacing SDP with metadata
   about media, displays and user controls.

   RTP data functionality can also be moved to the application itself.
   NAT traversal and other functions are delegated to HIP.

Thanks,
Henry

--B_3360047400_16103206
Content-type: text/html;
	charset="ISO-8859-1"
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<HTML>
<HEAD>
<TITLE>SIP APIs for Communications on the Web</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>Following the discussions on the Dispatch list in February and also in pri=
vate, we have posted a new version of the I-D:<BR>
<BR>
<a href=3D"http://www.ietf.org/id/draft-sinnreich-sip-web-apis-01.txt">http:/=
/www.ietf.org/id/draft-sinnreich-sip-web-apis-01.txt</a><BR>
<BR>
The clarifications include such topics as (in no particular order)<BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=
 STYLE=3D'font-size:13pt'>SIP and/or (vs.) XMPP/Jingle
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>HTML5 support for application scripts=20
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>Bidirectional HTTP, </SPAN></FONT><SPAN STYLE=3D'font-size=
:13pt'><FONT FACE=3D"Consolas, Courier New, Courier">IETF &quot;HyBi&quot; WG
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Consolas, Courie=
r New, Courier">Hiding SIP complexity under the right, standards based API
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">Drawing the line between basic usage scenarios for a sim=
ple API and yet how to support complex VoIP call flows
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">The SIP state machine moves to the business logic in UA =
and Web feature servers; which is preferable
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">The traffic and time spent with web applications far sur=
passes the time to talk on the phone (exceptions)
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">Added references for metadata standards to replace SDP
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">SDPng was already proposed in the early &#8216;90s
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">Keep only UDP for media transport, RTP data about the me=
dia is moved to the application as well
</FONT></SPAN><LI><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdana=
, Helvetica, Arial">New reference for HIP specific NAT traversal utilities.<=
BR>
</FONT></SPAN></UL><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri, Verdan=
a, Helvetica, Arial"><BR>
Please let us know how you feel these and other issues have been addressed.=
<BR>
<BR>
Abstract<BR>
<BR>
&nbsp;&nbsp;&nbsp;This memo describes a standards based approach to enable =
web based<BR>
&nbsp;&nbsp;&nbsp;interactive multimedia communications. &nbsp;The objectiv=
e is giving web<BR>
&nbsp;&nbsp;&nbsp;developers the software tools to add communication widget=
s to web<BR>
&nbsp;&nbsp;&nbsp;pages.<BR>
<BR>
&nbsp;&nbsp;&nbsp;The proposed SIP API does not necessarily require SIP pro=
tocol<BR>
&nbsp;&nbsp;&nbsp;expertise by web developers for basic multimedia communic=
ations<BR>
&nbsp;&nbsp;&nbsp;though it can be extended to port complex VoIP services t=
o the web.<BR>
&nbsp;&nbsp;&nbsp;The SIP API can also support the transition from network<=
BR>
&nbsp;&nbsp;&nbsp;infrastructure based VoIP to rich web based communication=
s.<BR>
<BR>
&nbsp;&nbsp;&nbsp;The benefits of the formal REST architecture of the web a=
re extended<BR>
&nbsp;&nbsp;&nbsp;to real time communications.<BR>
<BR>
&nbsp;&nbsp;&nbsp;Only two standard application layer protocols are require=
d: HTTP for<BR>
&nbsp;&nbsp;&nbsp;signaling data communications such as in SIP and/or XMPP =
and UDP for<BR>
&nbsp;&nbsp;&nbsp;real time media transport. We consider replacing SDP with=
 metadata<BR>
&nbsp;&nbsp;&nbsp;about media, displays and user controls.<BR>
<BR>
&nbsp;&nbsp;&nbsp;RTP data functionality can also be moved to the applicati=
on itself.<BR>
&nbsp;&nbsp;&nbsp;NAT traversal and other functions are delegated to HIP.<B=
R>
<BR>
Thanks,<BR>
Henry</FONT></SPAN>
</BODY>
</HTML>


--B_3360047400_16103206--



From R.Jesske@telekom.de  Wed Jun 23 00:33:55 2010
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From: <R.Jesske@telekom.de>
To: <john.elwell@siemens-enterprise.com>, <dispatch@ietf.org>
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Hi John,
After reading this thread again, I would like to propose two =
requirements as follows:

REQ-1:
It should be possible to support PSTN-SIP-PSTN scenarios where the =
reason of a call release can be transferred though the SIP domain =
without any loss of information and no change of reason.</t

REQ-2:
It must be possible to provide an accurate indication to a UAC, so that =
the UAC can provide a suitable indication=20
to the user.



As an additional Option there could be a third one as follows:

REQ-3:

UA may have the ability to display ISUP specific release causes or show =
a equivalent text.=20


Please indicate if you see this RE-3 within the scope or if it is =
already covered by REQ-2.

Opinions.=20

If people are happy I will provide an updated draft.

Best Regards

Roland



Deutsche Telekom Netzproduktion GmbH
Fixed Mobile Engineering Deutschland
Roland Jesske
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
+49 6151 628-2766 (Tel.)
+49 521 9210-3753  (Fax)
+49 171 8618-445  (Mobil)
E-Mail: r.jesske@telekom.de
www.telekom.com =20

Erleben, was verbindet.=20

Deutsche Telekom Netzproduktion GmbH
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr. DE 814645262

Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und nicht =
jede E-Mail drucken.


-----Urspr=FCngliche Nachricht-----
Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
Gesendet: Dienstag, 15. Juni 2010 16:28
An: Jesske, Roland; dispatch@ietf.org
Betreff: RE: Reason In =
responses(draft-jesske-dispatch-reason-in-responses-02)

=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: 07 June 2010 06:59
> To: Elwell, John; dispatch@ietf.org
> Subject: AW: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Hi John,
> Thank you for your comments:
> Here are my view.
>=20
> REQ-1:
> I think that REQ-1 is still a justification, because not each=20
> Operator will use SIP-T for tunnelling purposes.
> If you pass an untrusted domain SIP-T should not part,=20
> because it could include restricted numbers or other=20
> information that should not be presented to untrusted networks.
>=20
> Also only for passing the Q.850 Cause code to encapsulate the=20
> whole ISUP is a little over dimensioned.
> The SIP-T MIME can be lost due to the interdomain trust=20
> relation ships which the Reason header cannot.
>=20
> For in an multi carrier/service provider environment like=20
> Germany is, it will not be secure that each operator will=20
> support SIP-T.=20
[JRE] This helps to explain why SIP-T is not sufficient. IF SIP-T =
requires you to include additional information that you may not want to =
include when going outside a domain, that would be a reason for =
requiring a different solution. This was not made clear before.

John

>=20
> REQ-2
> It must be possible to provide an accurate indication to a=20
> UAC, so that the UAC can provide a suitable indication to the=20
> user.
>=20
> So that is the proposal for REQ-2.
>=20
> REQ-3=20
>    A UA may have the ability to display ISUP specific release=20
> causes or
>    show a equivalent text.
>=20
> So that would be sufficient enough for REQ-3. If people would=20
> like to see such text.
>=20
> REQ-3/4
> Your comment with regard to the type of UA is correct. I=20
> tried to satisfy people. The fact is that is not intended to=20
> include the Q.850 cause by an SIP end device. So the=20
> conclusion is changing REQ-3 and deleting REQ-4.
>=20
> So now I would like to ask the people which requirements they=20
> would like to see in the draft.
> As I understand John he sees only one (REQ-2)
>=20
> I propose REQ-1, -2 and -3.
>=20
> Opinions?
>=20
> Thank you and Best Regards
>=20
> Roland
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Zentrum Technik Einf=FChrung
> Roland Jesske
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628-2766 (Tel.)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobil)
> E-Mail: mailto:r.jesske@telekom.de
> http://www.telekom.com =20
>=20
> Erleben, was verbindet.=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> Albert Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> nicht jede E-Mail drucken.
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> Gesendet: Dienstag, 18. Mai 2010 21:37
> An: Jesske, Roland; dispatch@ietf.org
> Betreff: RE: Reason In=20
> responses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Roland,=20
>=20
> > -----Original Message-----
> > From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> > Sent: 18 May 2010 15:57
> > To: Elwell, John; dispatch@ietf.org
> > Subject: AW: Reason In=20
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >=20
> > Hi John,
> > Thank you for your comments.=20
> > It was asked to split the draft into requirements and at=20
> > least procedures.
> >=20
> > You are right that within the requirements draft is more that=20
> > requirements. It is also the motivation, justification and=20
> > examples. I can change the title if people are happy with it.=20
> > Question is if such a draft is also needed as RFC or if it=20
> > only seen as a temporary document during the discussion.
> [JRE] I was not proposing a title change, and I have no=20
> problem combining motivation with requirements, although I do=20
> feel the examples look too much like solution and are=20
> unnecessary. Although some people asked you to split the=20
> draft, in my opinion it was wasted effort, but it is done now.
>=20
> >=20
> > With regard to REQ-1 I propose the following re-wording:
> >=20
> > It should be possible to support within native SIP=20
> > environment PSTN-SIP-PSTN scenarios where the reason of a=20
> > call release can be transferred though the SIP domain
> > without any loss of information and no change of reason.
> >=20
> > I think this was discussed lengthy during the last meeting=20
> > and email discussion.
> > Not each PSTN GW will support ISUP MIME encapsulation. At=20
> > least the 3GPP IMS has no SIP. The IMS is not using SIP-T.
> [JRE] I don't think the new wording makes any difference.=20
> SIP-T is a means for tunnelling a Q.850 cause through a=20
> native SIP domain. The IETF already defines SIP-T as the=20
> solution, and if this solution is not sufficient, you need to=20
> cite reasons why not and identify a requirement that cannot=20
> be met using SIP-T. Just saying that IMS does not use SIP-T=20
> doesn't sound like a sufficient reason to me, because clearly=20
> IMS could be made to use SIP-T.
>=20
> To be clear, I am not saying that the Reason header field=20
> should not be used in this situation, if the header field is=20
> defined for use in responses in general in order meet other=20
> requirements. I am simply saying that REQ-1 is not sufficient=20
> justification for allowing the Reason header field in=20
> responses, since REQ-1 can be satisfied using an existing mechanism.
>=20
>=20
> >=20
> > REQ-2 and REQ-3
> > Yes it looks similar.
> >=20
> > REQ-3 was added because it was seen from user device=20
> > perspective that the Device have the possibility to show the=20
> > Q.850 cause but that such a End Device does not include Q.850=20
> > causes. Of course a MGC should have this possibility.
> >=20
> > I would be happy to shrink REQ-2 and 3 to your proposed text.
> >=20
> > REQ-2
> > It must be possible to provide an accurate indication to a=20
> > UAC, so that the UAC can provide a suitable indication to the=20
> > user. Whether this be an announcement, a textual message, an=20
> > icon, a flashing lamp, a tone, a vibration, an odour or=20
> > whatever, is a user interface matter.
> [JRE] I did not intend the second sentence to be part of the=20
> requirement - it was just for illustration of why the=20
> existing text, which focused on textual display, was inadequate.
>=20
> >=20
> > REQ-4 was the consequence of REQ-3 where we do not allow the=20
> > inclusion of a Q850 cause by a end device.
> >=20
> > So the trick is not allow the SIP end device to include a=20
> > Q850 cause and in some certain cases to allow an service=20
> > provider controlled application server to include a Reason=20
> > header with a Q.850 cause. But it should not be done=20
> > generally by SIP applications. So normally a ISUP=20
> > implementation will use ISUP causes but not SIP implementations.
> > But there are SIP Application servers out in the world that=20
> > will have a ISUP application (or ISUP like application that=20
> > reacts and behaves as within a ISUP network).
> [JRE] I found REQ-3 rather confusing. On re-reading it, the=20
> first sentence is a requirement concerning reception, and the=20
> second sentence seems (if I understand correctly) to be a=20
> statement about sending. Assuming my understanding is=20
> correct, making a statement, within a requirement, that=20
> something is out of scope seems to suggest it is not part of=20
> the requirement.
>=20
> In addition, the way SIP is specified, there is no real=20
> distinction between different types of UA. In other words,=20
> SIP procedures for UAs apply to UAs in general, not to=20
> specific types of UA like ASs. So writing a requirement that=20
> tries to distinguish between different types of UA does not=20
> seem to make sense.
>=20
> >=20
> > Question is now how to proceed further.
> >=20
> > Are people happy to have the documents split or put again=20
> > together with striking out the examples ect?
> > Or to have the documents as they are with some transfer of=20
> > text to the draft-jesske-dispatch-reason-in-responses-02 draft?
> [JRE] I don't have a strong opinion about document structure,=20
> although my personal preference would have been to leave it=20
> as one document, as it was originally. My main concern is=20
> trying to come up with a meaningful set of requirements, and=20
> in my opinion it reduces to a single requirement - being able=20
> to convey a Q.850 cause from PSTN (or similar) to a UAC so=20
> that the UAC can render meaningful information to the user.
>=20
> John
>=20
>=20
> >=20
> > Comments?
> >=20
> > Thank you and Best Regards
> >=20
> > Roland
> >=20
> > -----Urspr=FCngliche Nachricht-----
> > Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
> > Gesendet: Donnerstag, 13. Mai 2010 09:24
> > An: Jesske, Roland; dispatch@ietf.org
> > Betreff: RE: Reason In=20
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >=20
> > Roland,
> >=20
> > In my view, there is a simple requirement to convey Q.850=20
> > causes in SIP response messages, so that the UAC will receive=20
> > a more accurate indication of the reason for rejection when=20
> > rejection comes from PSTN. The cause mapping table helps to=20
> > illustrate this. This is not rocket science, and I am=20
> > disappointed that we have not found a quick way of=20
> > dispatching this and just getting the work done. In my=20
> > opinion the split into separate documents was unnecessary=20
> > (take as an example the recently-published RFC 5876, which=20
> > combines motivation and solution in one document), but it has=20
> > been done now. Unfortunately I don't think it has moved us=20
> > any further forward.
> >=20
> > I don't think the requirements in section 3 are particularly=20
> > well expressed. REQ-1 can already be met by tunnelling in=20
> > accordance with SIP-T. I would not expect the IETF to provide=20
> > a new solution specifically for that requirement.
> >=20
> > I think REQ-2 and REQ-3 boil down to the same thing - it must=20
> > be possible to provide an accurate indication to a UAC, so=20
> > that the UAC can provide a suitable indication to the user.=20
> > Whether this be an announcement, a textual message, an icon,=20
> > a flashing lamp, a tone, a vibration, an odour or whatever,=20
> > is a user interface matter.
> >=20
> > I also find REQ-4 rather problematic. If we have a solution=20
> > for PSTN, we would also have a solution for some other form=20
> > of gateway into a domain that provides a "PSTN-like service".=20
> > Unfortunately, if you try to bring this out as a separate=20
> > requirement, it raises the question of what exactly is this=20
> > "PSTN-like service", and if it is somehow SIP-based, why it=20
> > needs to be "PSTN-like", and so on.
> >=20
> > Basically I think there is only a single requirement here -=20
> > provision of a correct indication to a UAC when rejection=20
> > comes from the PSTN. I think that is a reasonable requirement=20
> > and we should just go ahead and do it, rather like we did=20
> > with RFC 5876 when we extended the range of messages that=20
> > PAI/PPI could be used in.
> >=20
> > I find the statement in REQ-3 "A inclusion of [Q.850] causes=20
> > is out of scope." very strange. I thought the whole idea was=20
> > to convey Q.850 causes, so why say they are out of scope? I=20
> > assume this is some sort of typo.
> >=20
> > I think one of the consequences of deciding to split=20
> > requirements and solution into separate drafts is that the=20
> > requirements draft has to stick to motivation and=20
> > requirements. I find too much in the draft (particularly the=20
> > examples) that smells of solution.
> >=20
> > John
> >=20
> >=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org=20
> > > [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> R.Jesske@telekom.de
> > > Sent: 13 May 2010 03:05
> > > To: dispatch@ietf.org
> > > Subject: Re: [dispatch] Reason In responses=20
> > > (draft-jesske-dispatch-reason-in-responses-02)
> > >=20
> > > Dear all,=20
> > > I have asked for comments on the two drafts mentioned below.=20
> > > So far I didn't get any. Does that mean that people are happy=20
> > > now with the content and we could proceed with it?=20
> > >=20
> > > Thank you and Best Regrads=20
> > >=20
> > > Roland=20
> > >=20
> > >=20
> > > _____________________________________________=20
> > > Von:    Jesske, Roland =20
> > > Gesendet:       Donnerstag, 8. April 2010 09:46=20
> > > An:     dispatch@ietf.org=20
> > > Betreff:        Reason In responses=20
> > > (draft-jesske-dispatch-reason-in-responses-02)=20
> > >=20
> > > Dear all,=20
> > > During the discussion concerning the Reason header field in=20
> > > Responses I was asked to split the document into an=20
> > > requirements part and an protocol part.
> > >=20
> > > So please feel free to comment on the drafts.=20
> > >=20
> > > http://64.170.98.42/html/draft-jesske-dispatch-reason-in-respo
> > > nses-02=20
> > > <http://64.170.98.42/html/draft-jesske-dispatch-reason-in-resp
> > > onses-02> =20
> > >=20
> > > http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-r
> > > esponses-00=20
> > > <http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-
> > > responses-00> =20
> > >=20
> > > Best regards=20
> > >=20
> > > Roland Jesske=20
> > >=20
> > >=20
> > > Deutsche Telekom Netzproduktion GmbH=20
> > > Zentrum Technik Einf=FChrung=20
> > > Roland Jesske=20
> > > Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
> > > +49 6151 628-2766 (Tel.)=20
> > > +49 521 9210-3753  (Fax)=20
> > > +49 171 8618-445  (Mobil)=20
> > > E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de> =20
> > > www.telekom.com <http:www.telekom.com>  =20
> > >=20
> > > Erleben, was verbindet.=20
> > >=20
> > > Deutsche Telekom Netzproduktion GmbH=20
> > > Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)=20
> > > Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> > > Albert Matheis, Klaus Peren=20
> > > Handelsregister: Amtsgericht Bonn HRB 14190=20
> > > Sitz der Gesellschaft: Bonn=20
> > > USt-IdNr. DE 814645262=20
> > >=20
> > > Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> > > nicht jede E-Mail drucken.=20
> > >=20
> > >=20
> > >=20
> >=20
>=20

From john.elwell@siemens-enterprise.com  Wed Jun 23 05:28:41 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 23 Jun 2010 14:28:43 +0200
Thread-Topic: Reason In responses(draft-jesske-dispatch-reason-in-responses-02)
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Subject: Re: [dispatch] Reason In responses(draft-jesske-dispatch-reason-in-responses-02)
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> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]
> Sent: 23 June 2010 08:33
> To: Elwell, John; dispatch@ietf.org
> Subject: AW: Reason In
> responses(draft-jesske-dispatch-reason-in-responses-02)
>
> Hi John,
> After reading this thread again, I would like to propose two
> requirements as follows:
>
> REQ-1:
> It should be possible to support PSTN-SIP-PSTN scenarios
> where the reason of a call release can be transferred though
> the SIP domain without any loss of information and no change
> of reason.</t
>
> REQ-2:
> It must be possible to provide an accurate indication to a
> UAC, so that the UAC can provide a suitable indication
> to the user.
>
>
>
> As an additional Option there could be a third one as follows:
>
> REQ-3:
>
> UA may have the ability to display ISUP specific release
> causes or show a equivalent text.
[JRE] This has no impact on the protocol and therefore is out of scope.

John


>
>
> Please indicate if you see this RE-3 within the scope or if
> it is already covered by REQ-2.
>
> Opinions.
>
> If people are happy I will provide an updated draft.
>
> Best Regards
>
> Roland
>
>
>
> Deutsche Telekom Netzproduktion GmbH
> Fixed Mobile Engineering Deutschland
> Roland Jesske
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628-2766 (Tel.)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobil)
> E-Mail: r.jesske@telekom.de
> www.telekom.com
>
> Erleben, was verbindet.
>
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),
> Albert Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>
> Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und
> nicht jede E-Mail drucken.
>
>
> -----Urspr=FCngliche Nachricht-----
> Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> Gesendet: Dienstag, 15. Juni 2010 16:28
> An: Jesske, Roland; dispatch@ietf.org
> Betreff: RE: Reason In
> responses(draft-jesske-dispatch-reason-in-responses-02)
>
>
>
> > -----Original Message-----
> > From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]
> > Sent: 07 June 2010 06:59
> > To: Elwell, John; dispatch@ietf.org
> > Subject: AW: Reason In
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >
> > Hi John,
> > Thank you for your comments:
> > Here are my view.
> >
> > REQ-1:
> > I think that REQ-1 is still a justification, because not each
> > Operator will use SIP-T for tunnelling purposes.
> > If you pass an untrusted domain SIP-T should not part,
> > because it could include restricted numbers or other
> > information that should not be presented to untrusted networks.
> >
> > Also only for passing the Q.850 Cause code to encapsulate the
> > whole ISUP is a little over dimensioned.
> > The SIP-T MIME can be lost due to the interdomain trust
> > relation ships which the Reason header cannot.
> >
> > For in an multi carrier/service provider environment like
> > Germany is, it will not be secure that each operator will
> > support SIP-T.
> [JRE] This helps to explain why SIP-T is not sufficient. IF
> SIP-T requires you to include additional information that you
> may not want to include when going outside a domain, that
> would be a reason for requiring a different solution. This
> was not made clear before.
>
> John
>
> >
> > REQ-2
> > It must be possible to provide an accurate indication to a
> > UAC, so that the UAC can provide a suitable indication to the
> > user.
> >
> > So that is the proposal for REQ-2.
> >
> > REQ-3
> >    A UA may have the ability to display ISUP specific release
> > causes or
> >    show a equivalent text.
> >
> > So that would be sufficient enough for REQ-3. If people would
> > like to see such text.
> >
> > REQ-3/4
> > Your comment with regard to the type of UA is correct. I
> > tried to satisfy people. The fact is that is not intended to
> > include the Q.850 cause by an SIP end device. So the
> > conclusion is changing REQ-3 and deleting REQ-4.
> >
> > So now I would like to ask the people which requirements they
> > would like to see in the draft.
> > As I understand John he sees only one (REQ-2)
> >
> > I propose REQ-1, -2 and -3.
> >
> > Opinions?
> >
> > Thank you and Best Regards
> >
> > Roland
> >
> >
> > Deutsche Telekom Netzproduktion GmbH
> > Zentrum Technik Einf=FChrung
> > Roland Jesske
> > Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> > +49 6151 628-2766 (Tel.)
> > +49 521 9210-3753  (Fax)
> > +49 171 8618-445  (Mobil)
> > E-Mail: mailto:r.jesske@telekom.de
> > http://www.telekom.com
> >
> > Erleben, was verbindet.
> >
> > Deutsche Telekom Netzproduktion GmbH
> > Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> > Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),
> > Albert Matheis, Klaus Peren
> > Handelsregister: Amtsgericht Bonn HRB 14190
> > Sitz der Gesellschaft: Bonn
> > USt-IdNr. DE 814645262
> >
> > Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und
> > nicht jede E-Mail drucken.
> >
> > -----Urspr=FCngliche Nachricht-----
> > Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > Gesendet: Dienstag, 18. Mai 2010 21:37
> > An: Jesske, Roland; dispatch@ietf.org
> > Betreff: RE: Reason In
> > responses(draft-jesske-dispatch-reason-in-responses-02)
> >
> > Roland,
> >
> > > -----Original Message-----
> > > From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]
> > > Sent: 18 May 2010 15:57
> > > To: Elwell, John; dispatch@ietf.org
> > > Subject: AW: Reason In
> > > responses(draft-jesske-dispatch-reason-in-responses-02)
> > >
> > > Hi John,
> > > Thank you for your comments.
> > > It was asked to split the draft into requirements and at
> > > least procedures.
> > >
> > > You are right that within the requirements draft is more that
> > > requirements. It is also the motivation, justification and
> > > examples. I can change the title if people are happy with it.
> > > Question is if such a draft is also needed as RFC or if it
> > > only seen as a temporary document during the discussion.
> > [JRE] I was not proposing a title change, and I have no
> > problem combining motivation with requirements, although I do
> > feel the examples look too much like solution and are
> > unnecessary. Although some people asked you to split the
> > draft, in my opinion it was wasted effort, but it is done now.
> >
> > >
> > > With regard to REQ-1 I propose the following re-wording:
> > >
> > > It should be possible to support within native SIP
> > > environment PSTN-SIP-PSTN scenarios where the reason of a
> > > call release can be transferred though the SIP domain
> > > without any loss of information and no change of reason.
> > >
> > > I think this was discussed lengthy during the last meeting
> > > and email discussion.
> > > Not each PSTN GW will support ISUP MIME encapsulation. At
> > > least the 3GPP IMS has no SIP. The IMS is not using SIP-T.
> > [JRE] I don't think the new wording makes any difference.
> > SIP-T is a means for tunnelling a Q.850 cause through a
> > native SIP domain. The IETF already defines SIP-T as the
> > solution, and if this solution is not sufficient, you need to
> > cite reasons why not and identify a requirement that cannot
> > be met using SIP-T. Just saying that IMS does not use SIP-T
> > doesn't sound like a sufficient reason to me, because clearly
> > IMS could be made to use SIP-T.
> >
> > To be clear, I am not saying that the Reason header field
> > should not be used in this situation, if the header field is
> > defined for use in responses in general in order meet other
> > requirements. I am simply saying that REQ-1 is not sufficient
> > justification for allowing the Reason header field in
> > responses, since REQ-1 can be satisfied using an existing mechanism.
> >
> >
> > >
> > > REQ-2 and REQ-3
> > > Yes it looks similar.
> > >
> > > REQ-3 was added because it was seen from user device
> > > perspective that the Device have the possibility to show the
> > > Q.850 cause but that such a End Device does not include Q.850
> > > causes. Of course a MGC should have this possibility.
> > >
> > > I would be happy to shrink REQ-2 and 3 to your proposed text.
> > >
> > > REQ-2
> > > It must be possible to provide an accurate indication to a
> > > UAC, so that the UAC can provide a suitable indication to the
> > > user. Whether this be an announcement, a textual message, an
> > > icon, a flashing lamp, a tone, a vibration, an odour or
> > > whatever, is a user interface matter.
> > [JRE] I did not intend the second sentence to be part of the
> > requirement - it was just for illustration of why the
> > existing text, which focused on textual display, was inadequate.
> >
> > >
> > > REQ-4 was the consequence of REQ-3 where we do not allow the
> > > inclusion of a Q850 cause by a end device.
> > >
> > > So the trick is not allow the SIP end device to include a
> > > Q850 cause and in some certain cases to allow an service
> > > provider controlled application server to include a Reason
> > > header with a Q.850 cause. But it should not be done
> > > generally by SIP applications. So normally a ISUP
> > > implementation will use ISUP causes but not SIP implementations.
> > > But there are SIP Application servers out in the world that
> > > will have a ISUP application (or ISUP like application that
> > > reacts and behaves as within a ISUP network).
> > [JRE] I found REQ-3 rather confusing. On re-reading it, the
> > first sentence is a requirement concerning reception, and the
> > second sentence seems (if I understand correctly) to be a
> > statement about sending. Assuming my understanding is
> > correct, making a statement, within a requirement, that
> > something is out of scope seems to suggest it is not part of
> > the requirement.
> >
> > In addition, the way SIP is specified, there is no real
> > distinction between different types of UA. In other words,
> > SIP procedures for UAs apply to UAs in general, not to
> > specific types of UA like ASs. So writing a requirement that
> > tries to distinguish between different types of UA does not
> > seem to make sense.
> >
> > >
> > > Question is now how to proceed further.
> > >
> > > Are people happy to have the documents split or put again
> > > together with striking out the examples ect?
> > > Or to have the documents as they are with some transfer of
> > > text to the draft-jesske-dispatch-reason-in-responses-02 draft?
> > [JRE] I don't have a strong opinion about document structure,
> > although my personal preference would have been to leave it
> > as one document, as it was originally. My main concern is
> > trying to come up with a meaningful set of requirements, and
> > in my opinion it reduces to a single requirement - being able
> > to convey a Q.850 cause from PSTN (or similar) to a UAC so
> > that the UAC can render meaningful information to the user.
> >
> > John
> >
> >
> > >
> > > Comments?
> > >
> > > Thank you and Best Regards
> > >
> > > Roland
> > >
> > > -----Urspr=FCngliche Nachricht-----
> > > Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> > > Gesendet: Donnerstag, 13. Mai 2010 09:24
> > > An: Jesske, Roland; dispatch@ietf.org
> > > Betreff: RE: Reason In
> > > responses(draft-jesske-dispatch-reason-in-responses-02)
> > >
> > > Roland,
> > >
> > > In my view, there is a simple requirement to convey Q.850
> > > causes in SIP response messages, so that the UAC will receive
> > > a more accurate indication of the reason for rejection when
> > > rejection comes from PSTN. The cause mapping table helps to
> > > illustrate this. This is not rocket science, and I am
> > > disappointed that we have not found a quick way of
> > > dispatching this and just getting the work done. In my
> > > opinion the split into separate documents was unnecessary
> > > (take as an example the recently-published RFC 5876, which
> > > combines motivation and solution in one document), but it has
> > > been done now. Unfortunately I don't think it has moved us
> > > any further forward.
> > >
> > > I don't think the requirements in section 3 are particularly
> > > well expressed. REQ-1 can already be met by tunnelling in
> > > accordance with SIP-T. I would not expect the IETF to provide
> > > a new solution specifically for that requirement.
> > >
> > > I think REQ-2 and REQ-3 boil down to the same thing - it must
> > > be possible to provide an accurate indication to a UAC, so
> > > that the UAC can provide a suitable indication to the user.
> > > Whether this be an announcement, a textual message, an icon,
> > > a flashing lamp, a tone, a vibration, an odour or whatever,
> > > is a user interface matter.
> > >
> > > I also find REQ-4 rather problematic. If we have a solution
> > > for PSTN, we would also have a solution for some other form
> > > of gateway into a domain that provides a "PSTN-like service".
> > > Unfortunately, if you try to bring this out as a separate
> > > requirement, it raises the question of what exactly is this
> > > "PSTN-like service", and if it is somehow SIP-based, why it
> > > needs to be "PSTN-like", and so on.
> > >
> > > Basically I think there is only a single requirement here -
> > > provision of a correct indication to a UAC when rejection
> > > comes from the PSTN. I think that is a reasonable requirement
> > > and we should just go ahead and do it, rather like we did
> > > with RFC 5876 when we extended the range of messages that
> > > PAI/PPI could be used in.
> > >
> > > I find the statement in REQ-3 "A inclusion of [Q.850] causes
> > > is out of scope." very strange. I thought the whole idea was
> > > to convey Q.850 causes, so why say they are out of scope? I
> > > assume this is some sort of typo.
> > >
> > > I think one of the consequences of deciding to split
> > > requirements and solution into separate drafts is that the
> > > requirements draft has to stick to motivation and
> > > requirements. I find too much in the draft (particularly the
> > > examples) that smells of solution.
> > >
> > > John
> > >
> > >
> > > > -----Original Message-----
> > > > From: dispatch-bounces@ietf.org
> > > > [mailto:dispatch-bounces@ietf.org] On Behalf Of
> > R.Jesske@telekom.de
> > > > Sent: 13 May 2010 03:05
> > > > To: dispatch@ietf.org
> > > > Subject: Re: [dispatch] Reason In responses
> > > > (draft-jesske-dispatch-reason-in-responses-02)
> > > >
> > > > Dear all,
> > > > I have asked for comments on the two drafts mentioned below.
> > > > So far I didn't get any. Does that mean that people are happy
> > > > now with the content and we could proceed with it?
> > > >
> > > > Thank you and Best Regrads
> > > >
> > > > Roland
> > > >
> > > >
> > > > _____________________________________________
> > > > Von:    Jesske, Roland
> > > > Gesendet:       Donnerstag, 8. April 2010 09:46
> > > > An:     dispatch@ietf.org
> > > > Betreff:        Reason In responses
> > > > (draft-jesske-dispatch-reason-in-responses-02)
> > > >
> > > > Dear all,
> > > > During the discussion concerning the Reason header field in
> > > > Responses I was asked to split the document into an
> > > > requirements part and an protocol part.
> > > >
> > > > So please feel free to comment on the drafts.
> > > >
> > > > http://64.170.98.42/html/draft-jesske-dispatch-reason-in-respo
> > > > nses-02
> > > > <http://64.170.98.42/html/draft-jesske-dispatch-reason-in-resp
> > > > onses-02>
> > > >
> > > > http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-r
> > > > esponses-00
> > > > <http://64.170.98.42/html/draft-jesske-dispatch-req-reason-in-
> > > > responses-00>
> > > >
> > > > Best regards
> > > >
> > > > Roland Jesske
> > > >
> > > >
> > > > Deutsche Telekom Netzproduktion GmbH
> > > > Zentrum Technik Einf=FChrung
> > > > Roland Jesske
> > > > Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> > > > +49 6151 628-2766 (Tel.)
> > > > +49 521 9210-3753  (Fax)
> > > > +49 171 8618-445  (Mobil)
> > > > E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de>
> > > > www.telekom.com <http:www.telekom.com>
> > > >
> > > > Erleben, was verbindet.
> > > >
> > > > Deutsche Telekom Netzproduktion GmbH
> > > > Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> > > > Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),
> > > > Albert Matheis, Klaus Peren
> > > > Handelsregister: Amtsgericht Bonn HRB 14190
> > > > Sitz der Gesellschaft: Bonn
> > > > USt-IdNr. DE 814645262
> > > >
> > > > Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und
> > > > nicht jede E-Mail drucken.
> > > >
> > > >
> > > >
> > >
> >
>

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To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "john.elwell@siemens-enterprise.com" <john.elwell@siemens-enterprise.com>,  "dispatch@ietf.org" <dispatch@ietf.org>
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Subject: Re: [dispatch] Reason In	responses(draft-jesske-dispatch-reason-in-responses-02)
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of R.=
Jesske@telekom.de [R.Jesske@telekom.de]

Hi John,
After reading this thread again, I would like to propose two requirements a=
s follows:

REQ-1:
It should be possible to support PSTN-SIP-PSTN scenarios where the reason o=
f a call release can be transferred though the SIP domain without any loss =
of information and no change of reason.</t

REQ-2:
It must be possible to provide an accurate indication to a UAC, so that the=
 UAC can provide a suitable indication
to the user.

REQ-3:
UA may have the ability to display ISUP specific release causes or show a e=
quivalent text.
_______________________________________________

For REQ-1, I would rephrase it to be something like "where an ISUP release =
code can be transferred transparently through the SIP domain".  Both "loss =
of information" and "no change of reason" are subject to interpretation -- =
what we're really hunting for is the ISUP release code (as a *number*) to g=
o in one end and come out the other end unchanged.

As written REQ-2 is rather vague.  What does "accurate" mean in this contex=
t?

I think what you mean by REQ-3 is "A SIP UAC originating a dialog which ter=
minates at a SIP-to-PSTN gateway can determine the exact ISUP release code =
provided by the gateway."

Dale

From tom111.taylor@bell.net  Wed Jun 23 09:37:40 2010
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Date: Wed, 23 Jun 2010 12:37:43 -0400
From: Tom Taylor <tom111.taylor@bell.net>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Reason In	responses(draft-jesske-dispatch-reason-in-responses-02)
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Query on REQ-3: are we talking about ISUP or ISDN release causes being forwarded 
from the PSTN?

R.Jesske@telekom.de wrote:
> Hi John,
> After reading this thread again, I would like to propose two requirements as follows:
> 
> REQ-1:
> It should be possible to support PSTN-SIP-PSTN scenarios where the reason of a call release can be transferred though the SIP domain without any loss of information and no change of reason.</t
> 
> REQ-2:
> It must be possible to provide an accurate indication to a UAC, so that the UAC can provide a suitable indication 
> to the user.
> 
> 
> 
> As an additional Option there could be a third one as follows:
> 
> REQ-3:
> 
> UA may have the ability to display ISUP specific release causes or show a equivalent text. 
> 
> 
> Please indicate if you see this RE-3 within the scope or if it is already covered by REQ-2.
> 
> Opinions. 
> 
> If people are happy I will provide an updated draft.
> 
> Best Regards
> 
> Roland
> 
...

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From: "DOLLY, MARTIN C (ATTLABS)" <md3135@att.com>
To: "Tom Taylor" <tom111.taylor@bell.net>, <R.Jesske@telekom.de>
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Subject: Re: [dispatch] ReasonIn	responses(draft-jesske-dispatch-reason-in-responses-02)
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ISUP, though there is an overlap with ISDN

If from ISDN, then there is an interworking ISDN-->ISUP-->SIP

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Tom Taylor
Sent: Wednesday, June 23, 2010 12:38 PM
To: R.Jesske@telekom.de
Cc: dispatch@ietf.org
Subject: Re: [dispatch] ReasonIn
responses(draft-jesske-dispatch-reason-in-responses-02)

Query on REQ-3: are we talking about ISUP or ISDN release causes being
forwarded=20
from the PSTN?

R.Jesske@telekom.de wrote:
> Hi John,
> After reading this thread again, I would like to propose two
requirements as follows:
>=20
> REQ-1:
> It should be possible to support PSTN-SIP-PSTN scenarios where the
reason of a call release can be transferred though the SIP domain
without any loss of information and no change of reason.</t
>=20
> REQ-2:
> It must be possible to provide an accurate indication to a UAC, so
that the UAC can provide a suitable indication=20
> to the user.
>=20
>=20
>=20
> As an additional Option there could be a third one as follows:
>=20
> REQ-3:
>=20
> UA may have the ability to display ISUP specific release causes or
show a equivalent text.=20
>=20
>=20
> Please indicate if you see this RE-3 within the scope or if it is
already covered by REQ-2.
>=20
> Opinions.=20
>=20
> If people are happy I will provide an updated draft.
>=20
> Best Regards
>=20
> Roland
>=20
...
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: "DOLLY, MARTIN C (ATTLABS)" <md3135@att.com>, Tom Taylor <tom111.taylor@bell.net>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of DO=
LLY, MARTIN C (ATTLABS) [md3135@att.com]

ISUP, though there is an overlap with ISDN

If from ISDN, then there is an interworking ISDN-->ISUP-->SIP
_______________________________________________

One thing I haven't seen discussed is what is the meaning of carrying the I=
SUP release code transparently if there is more than one variety of ISUP, i=
f the same release code has different meanings in different PSTN networks. =
 In a perfect world, SIP would carry not only the ISUP release code, but so=
me identification of the *system* of ISUP release codes that it was drawn f=
rom.

Dale

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From: <R.Jesske@telekom.de>
To: <dworley@avaya.com>, <john.elwell@siemens-enterprise.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Reason Inresponses(draft-jesske-dispatch-reason-in-responses-02)
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Hi Dale,
Perhaps some more words, like that:

REQ-2:
It must be possible to provide an accurate indication to a UAC why the =
call was released, so that the UAC can provide a suitable indication.=20

After discussion on this requirement the proposal was to keep it =
generic. So now I tried to add some words to make is a little more =
related to the kind of indication.

As it looks like you would like to see the 3rd requirement in addition. =
Correct?

If yes then I changed the wording you provided to:
REQ-3 :=20
A SIP UAC originating a dialog which terminates at a SIP-to-PSTN gateway =
can determine the exact ISDN/ISUP Q.850 release code interworked by the =
gateway or show an equivalent text.

The Question then is how to express it that the SIP Dialog of course =
terminates at the GW but the call itself is terminated within the =
PSTN/ISDN network. And the cause is provided by the PSTN/ISDN network. =
So that why I took the wording interworked by the gateway.

Other views?

Best Regards

Roland

Mit freundlichen Gr=FC=DFen
Roland Jesske


Deutsche Telekom Netzproduktion GmbH
Fixed Mobile Engineering Deutschland
Roland Jesske
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
+49 6151 628-2766 (Tel.)
+49 521 9210-3753  (Fax)
+49 171 8618-445  (Mobil)
E-Mail: r.jesske@telekom.de
www.telekom.com =20

Erleben, was verbindet.=20

Deutsche Telekom Netzproduktion GmbH
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr. DE 814645262

Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und nicht =
jede E-Mail drucken.


-----Urspr=FCngliche Nachricht-----
Von: WORLEY, Dale R (Dale) [mailto:dworley@avaya.com]=20
Gesendet: Mittwoch, 23. Juni 2010 18:31
An: Jesske, Roland; john.elwell@siemens-enterprise.com; =
dispatch@ietf.org
Betreff: RE: [dispatch] Reason =
Inresponses(draft-jesske-dispatch-reason-in-responses-02)

________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of =
R.Jesske@telekom.de [R.Jesske@telekom.de]

Hi John,
After reading this thread again, I would like to propose two =
requirements as follows:

REQ-1:
It should be possible to support PSTN-SIP-PSTN scenarios where the =
reason of a call release can be transferred though the SIP domain =
without any loss of information and no change of reason.</t

REQ-2:
It must be possible to provide an accurate indication to a UAC, so that =
the UAC can provide a suitable indication
to the user.

REQ-3:
UA may have the ability to display ISUP specific release causes or show =
a equivalent text.
_______________________________________________

For REQ-1, I would rephrase it to be something like "where an ISUP =
release code can be transferred transparently through the SIP domain".  =
Both "loss of information" and "no change of reason" are subject to =
interpretation -- what we're really hunting for is the ISUP release code =
(as a *number*) to go in one end and come out the other end unchanged.

As written REQ-2 is rather vague.  What does "accurate" mean in this =
context?

I think what you mean by REQ-3 is "A SIP UAC originating a dialog which =
terminates at a SIP-to-PSTN gateway can determine the exact ISUP release =
code provided by the gateway."

Dale

From john.elwell@siemens-enterprise.com  Thu Jun 24 02:00:50 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "dworley@avaya.com" <dworley@avaya.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 24 Jun 2010 11:00:55 +0200
Thread-Topic: [dispatch] Reason Inresponses(draft-jesske-dispatch-reason-in-responses-02)
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=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: 24 June 2010 08:44
> To: dworley@avaya.com; Elwell, John; dispatch@ietf.org
> Subject: AW: [dispatch] Reason=20
> Inresponses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Hi Dale,
> Perhaps some more words, like that:
>=20
> REQ-2:
> It must be possible to provide an accurate indication to a=20
> UAC why the call was released, so that the UAC can provide a=20
> suitable indication.=20
>=20
> After discussion on this requirement the proposal was to keep=20
> it generic. So now I tried to add some words to make is a=20
> little more related to the kind of indication.
>=20
> As it looks like you would like to see the 3rd requirement in=20
> addition. Correct?
>=20
> If yes then I changed the wording you provided to:
> REQ-3 :=20
> A SIP UAC originating a dialog which terminates at a=20
> SIP-to-PSTN gateway can determine the exact ISDN/ISUP Q.850=20
> release code interworked by the gateway or show an equivalent text.
[JRE] The words " or show an equivalent text" are user-interface specific a=
nd should be deleted. Determination of the Q.850 cause is sufficient - what=
 the UA then does with it (e.g., displays the cause number, displays text, =
shows an icon, plays a tone or announcement) is outside the scope of SIP.

I still have difficult determining the need for both REQ-2 and REQ-3. If we=
 have REQ-3, why do we need REQ-2. SIP already provides a means of satisfyi=
ng REQ-2 through regular SIP response codes, since REQ-2 says nothing about=
 information from PSTN. If we talk about Q.850 causes in REQ-2, what would =
be the difference between REQ-2 and REQ-3?

John



>=20
> The Question then is how to express it that the SIP Dialog of=20
> course terminates at the GW but the call itself is terminated=20
> within the PSTN/ISDN network. And the cause is provided by=20
> the PSTN/ISDN network. So that why I took the wording=20
> interworked by the gateway.
>=20
> Other views?
>=20
> Best Regards
>=20
> Roland
>=20
> Mit freundlichen Gr=FC=DFen
> Roland Jesske
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Fixed Mobile Engineering Deutschland
> Roland Jesske
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628-2766 (Tel.)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobil)
> E-Mail: r.jesske@telekom.de
> www.telekom.com =20
>=20
> Erleben, was verbindet.=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> Albert Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> nicht jede E-Mail drucken.
>=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: WORLEY, Dale R (Dale) [mailto:dworley@avaya.com]=20
> Gesendet: Mittwoch, 23. Juni 2010 18:31
> An: Jesske, Roland; john.elwell@siemens-enterprise.com;=20
> dispatch@ietf.org
> Betreff: RE: [dispatch] Reason=20
> Inresponses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org]=20
> On Behalf Of R.Jesske@telekom.de [R.Jesske@telekom.de]
>=20
> Hi John,
> After reading this thread again, I would like to propose two=20
> requirements as follows:
>=20
> REQ-1:
> It should be possible to support PSTN-SIP-PSTN scenarios=20
> where the reason of a call release can be transferred though=20
> the SIP domain without any loss of information and no change=20
> of reason.</t
>=20
> REQ-2:
> It must be possible to provide an accurate indication to a=20
> UAC, so that the UAC can provide a suitable indication
> to the user.
>=20
> REQ-3:
> UA may have the ability to display ISUP specific release=20
> causes or show a equivalent text.
> _______________________________________________
>=20
> For REQ-1, I would rephrase it to be something like "where an=20
> ISUP release code can be transferred transparently through=20
> the SIP domain".  Both "loss of information" and "no change=20
> of reason" are subject to interpretation -- what we're really=20
> hunting for is the ISUP release code (as a *number*) to go in=20
> one end and come out the other end unchanged.
>=20
> As written REQ-2 is rather vague.  What does "accurate" mean=20
> in this context?
>=20
> I think what you mean by REQ-3 is "A SIP UAC originating a=20
> dialog which terminates at a SIP-to-PSTN gateway can=20
> determine the exact ISUP release code provided by the gateway."
>=20
> Dale
> =

From R.Jesske@telekom.de  Thu Jun 24 05:43:54 2010
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Thread-Topic: [dispatch] Reason Inresponses(draft-jesske-dispatch-reason-in-responses-02)
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From: <R.Jesske@telekom.de>
To: <john.elwell@siemens-enterprise.com>, <dworley@avaya.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Reason Inresponses(draft-jesske-dispatch-reason-in-responses-02)
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Hi John,
>From my side we can go with two requirements. Your comment is correct =
and REQ-2 and 3 are overlapping.
=20
Who would like to see in addition REQ-3 please indicate and Who not. I'm =
waiting for your response.

Thank you.

Best Regards

Roland

-----Urspr=FCngliche Nachricht-----
Von: Elwell, John [mailto:john.elwell@siemens-enterprise.com]=20
Gesendet: Donnerstag, 24. Juni 2010 11:01
An: Jesske, Roland; dworley@avaya.com; dispatch@ietf.org
Betreff: RE: [dispatch] Reason =
Inresponses(draft-jesske-dispatch-reason-in-responses-02)

=20

> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]=20
> Sent: 24 June 2010 08:44
> To: dworley@avaya.com; Elwell, John; dispatch@ietf.org
> Subject: AW: [dispatch] Reason=20
> Inresponses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> Hi Dale,
> Perhaps some more words, like that:
>=20
> REQ-2:
> It must be possible to provide an accurate indication to a=20
> UAC why the call was released, so that the UAC can provide a=20
> suitable indication.=20
>=20
> After discussion on this requirement the proposal was to keep=20
> it generic. So now I tried to add some words to make is a=20
> little more related to the kind of indication.
>=20
> As it looks like you would like to see the 3rd requirement in=20
> addition. Correct?
>=20
> If yes then I changed the wording you provided to:
> REQ-3 :=20
> A SIP UAC originating a dialog which terminates at a=20
> SIP-to-PSTN gateway can determine the exact ISDN/ISUP Q.850=20
> release code interworked by the gateway or show an equivalent text.
[JRE] The words " or show an equivalent text" are user-interface =
specific and should be deleted. Determination of the Q.850 cause is =
sufficient - what the UA then does with it (e.g., displays the cause =
number, displays text, shows an icon, plays a tone or announcement) is =
outside the scope of SIP.

I still have difficult determining the need for both REQ-2 and REQ-3. If =
we have REQ-3, why do we need REQ-2. SIP already provides a means of =
satisfying REQ-2 through regular SIP response codes, since REQ-2 says =
nothing about information from PSTN. If we talk about Q.850 causes in =
REQ-2, what would be the difference between REQ-2 and REQ-3?

John



>=20
> The Question then is how to express it that the SIP Dialog of=20
> course terminates at the GW but the call itself is terminated=20
> within the PSTN/ISDN network. And the cause is provided by=20
> the PSTN/ISDN network. So that why I took the wording=20
> interworked by the gateway.
>=20
> Other views?
>=20
> Best Regards
>=20
> Roland
>=20
> Mit freundlichen Gr=FC=DFen
> Roland Jesske
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Fixed Mobile Engineering Deutschland
> Roland Jesske
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628-2766 (Tel.)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobil)
> E-Mail: r.jesske@telekom.de
> www.telekom.com =20
>=20
> Erleben, was verbindet.=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender),=20
> Albert Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und=20
> nicht jede E-Mail drucken.
>=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: WORLEY, Dale R (Dale) [mailto:dworley@avaya.com]=20
> Gesendet: Mittwoch, 23. Juni 2010 18:31
> An: Jesske, Roland; john.elwell@siemens-enterprise.com;=20
> dispatch@ietf.org
> Betreff: RE: [dispatch] Reason=20
> Inresponses(draft-jesske-dispatch-reason-in-responses-02)
>=20
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org]=20
> On Behalf Of R.Jesske@telekom.de [R.Jesske@telekom.de]
>=20
> Hi John,
> After reading this thread again, I would like to propose two=20
> requirements as follows:
>=20
> REQ-1:
> It should be possible to support PSTN-SIP-PSTN scenarios=20
> where the reason of a call release can be transferred though=20
> the SIP domain without any loss of information and no change=20
> of reason.</t
>=20
> REQ-2:
> It must be possible to provide an accurate indication to a=20
> UAC, so that the UAC can provide a suitable indication
> to the user.
>=20
> REQ-3:
> UA may have the ability to display ISUP specific release=20
> causes or show a equivalent text.
> _______________________________________________
>=20
> For REQ-1, I would rephrase it to be something like "where an=20
> ISUP release code can be transferred transparently through=20
> the SIP domain".  Both "loss of information" and "no change=20
> of reason" are subject to interpretation -- what we're really=20
> hunting for is the ISUP release code (as a *number*) to go in=20
> one end and come out the other end unchanged.
>=20
> As written REQ-2 is rather vague.  What does "accurate" mean=20
> in this context?
>=20
> I think what you mean by REQ-3 is "A SIP UAC originating a=20
> dialog which terminates at a SIP-to-PSTN gateway can=20
> determine the exact ISUP release code provided by the gateway."
>=20
> Dale
>=20

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Date: Thu, 24 Jun 2010 13:50:05 +0000 (GMT)
From: Ian Elz <ian_elz@yahoo.co.uk>
To: dispatch@ietf.org, "WORLEY, Dale R \(Dale\)" <dworley@avaya.com>
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Cc: "DOLLY, MARTIN C \(ATTLABS\)" <md3135@att.com>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
Subject: Re: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-responses-02)
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Dale,=0A=0AThe Q.850 cause values are standardised across the ISUP network.=
 As ISUP is used to interconnect the PSTN and PLMN worldwide the values are=
 universal.=0A=0AThe fact that the Reason header will indicate a protocol o=
f "Q.850" =0A=0AThe only thing that is not considered in this draft is that=
 there may be 'location' and 'diagnostic' information associated the cause =
values.=0A=0AThe Location indicates if the release cause comes from teh use=
r, private networl, public network, transit network etc.=0A=0AThe Diagnosti=
cs provides additional information as to why the realease occured and other=
 information which may be cause specific.=0A=0AThis additional information =
is not usually used and I don't believe that it is the intention of this dr=
aft to expand the Reason header to carry the additional information. Any in=
formation which is required will need to be mapped by other mechanism betwe=
en the ISUP and SIP networks, e.g. CCBS possible indicator.=0A=0AIan Elz=0A=
=0A=0A=0AFrom: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] =0AOn =
Behalf Of "WORLEY, Dale R (Dale)" <dworley@avaya.com>=0A=0A=0A_____________=
___________________=0A    From: dispatch-bounces@ietf.org [dispatch-bounces=
@ietf.org] =0AOn Behalf Of DOLLY, MARTIN C (ATTLABS) [md3135@att.com]=0A=0A=
    ISUP, though there is an overlap with ISDN=0A=0A    If from ISDN, then =
there is an interworking ISDN-->ISUP-->SIP=0A______________________________=
_________________=0A=0AOne thing I haven't seen discussed is what is the me=
aning of carrying the =0AISUP release code transparently if there is more t=
han one variety of =0AISUP, if the same release code has different meanings=
 in different PSTN =0Anetworks.  In a perfect world, SIP would carry not on=
ly the ISUP release code, but some identification of the *system* of ISUP r=
elease codes =0Athat it was drawn from.=0A=0ADale=0A=0A=0A      
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<html><head><style type=3D"text/css"><!-- DIV {margin:0px;} --></style></he=
ad><body><div style=3D"font-family:arial,helvetica,sans-serif;font-size:10p=
t"><div>Dale,<br><br>The Q.850 cause values are standardised across the ISU=
P network. As ISUP is used to interconnect the PSTN and PLMN worldwide the =
values are universal.<br><br>The fact that the Reason header will indicate =
a protocol of "Q.850" <br><br>The only thing that is not considered in this=
 draft is that there may be 'location' and 'diagnostic' information associa=
ted the cause values.<br><br>The Location indicates if the release cause co=
mes from teh user, private networl, public network, transit network etc.<br=
><br>The Diagnostics provides additional information as to why the realease=
 occured and other information which may be cause specific.<br><br>This add=
itional information is not usually used and I don't believe that it is the =
intention of this draft to expand the Reason header to carry the
 additional information. Any information which is required will need to be =
mapped by other mechanism between the ISUP and SIP networks, e.g. CCBS poss=
ible indicator.<br><br>Ian Elz<br></div><div style=3D"font-family: arial,he=
lvetica,sans-serif; font-size: 10pt;"><br><br>From: <a ymailto=3D"mailto:di=
spatch-bounces@ietf.org" href=3D"mailto:dispatch-bounces@ietf.org">dispatch=
-bounces@ietf.org</a> [<a ymailto=3D"mailto:dispatch-bounces@ietf.org" href=
=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a>] =0AOn =
Behalf Of "WORLEY, Dale R (Dale)" &lt;dworley@avaya.com&gt;<br><div style=
=3D"font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size=
=3D"2" face=3D"Tahoma"><hr size=3D"1"><b><span style=3D"font-weight: bold;"=
></span></b></font>&nbsp;&nbsp;&nbsp; From: <a ymailto=3D"mailto:dispatch-b=
ounces@ietf.org" href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces=
@ietf.org</a> [<a ymailto=3D"mailto:dispatch-bounces@ietf.org" href=3D"mail=
to:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a>] =0AOn Behalf O=
f DOLLY, MARTIN C (ATTLABS) [<a ymailto=3D"mailto:md3135@att.com" href=3D"m=
ailto:md3135@att.com">md3135@att.com</a>]<br><br>&nbsp;&nbsp;&nbsp; ISUP,=
=0A though there is an overlap with ISDN<br><br>&nbsp;&nbsp;&nbsp; If from =
ISDN, then there is=0A an interworking ISDN--&gt;ISUP--&gt;SIP<br>_________=
______________________________________<br><br>One=0A thing I haven't seen d=
iscussed is what is the meaning of carrying the =0AISUP release code transp=
arently if there is more than one variety of =0AISUP, if the same release c=
ode has different meanings in different PSTN =0Anetworks.&nbsp; In a perfec=
t world, SIP would carry not only the ISUP release=0A code, but some identi=
fication of the *system* of ISUP release codes =0Athat it was drawn from.<b=
r><br>Dale</div></div>=0A</div><br>=0A=0A=0A=0A      </body></html>
--0-651532606-1277387405=:82617--


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Subject: Re: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-responses-02)
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Hi Ian,
the location field for the Reason header was discussed in the past. But =
it was decided not to include to many information. My impression is that =
the location field is not used often for services or other network =
feature. The only value would be for O&M reasons where more information =
is given. So your analysis in the last sentence is correct.
=20
This draft is not extending the Reason header it is only specifying the =
use of the Reason Header within responses.
=20
Best Regards
=20
Roland

Mit freundlichen Gr=FC=DFen=20
Roland Jesske=20


Deutsche Telekom Netzproduktion GmbH=20
Fixed Mobile Engineering Deutschland=20
Roland Jesske=20
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt=20
+49 6151 628-2766 (Tel.)=20
+49 521 9210-3753  (Fax)=20
+49 171 8618-445  (Mobil)=20
E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de> =20
www.telekom.com <http:www.telekom.com>  =20

Erleben, was verbindet.=20

Deutsche Telekom Netzproduktion GmbH=20
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)=20
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren=20
Handelsregister: Amtsgericht Bonn HRB 14190=20
Sitz der Gesellschaft: Bonn=20
USt-IdNr. DE 814645262=20

Gro=DFe Ver=E4nderungen fangen klein an - Ressourcen schonen und nicht =
jede E-Mail drucken.=20

=20

________________________________

Von: Ian Elz [mailto:ian_elz@yahoo.co.uk]=20
Gesendet: Donnerstag, 24. Juni 2010 15:50
An: dispatch@ietf.org; WORLEY, Dale R (Dale)
Cc: DOLLY, MARTIN C (ATTLABS); Tom Taylor; Jesske, Roland
Betreff: Re: [dispatch] ReasonIn =
responses(draft-jesske-dispatch-reason-in-responses-02)


Dale,

The Q.850 cause values are standardised across the ISUP network. As ISUP =
is used to interconnect the PSTN and PLMN worldwide the values are =
universal.

The fact that the Reason header will indicate a protocol of "Q.850"=20

The only thing that is not considered in this draft is that there may be =
'location' and 'diagnostic' information associated the cause values.

The Location indicates if the release cause comes from teh user, private =
networl, public network, transit network etc.

The Diagnostics provides additional information as to why the realease =
occured and other information which may be cause specific.

This additional information is not usually used and I don't believe that =
it is the intention of this draft to expand the Reason header to carry =
the additional information. Any information which is required will need =
to be mapped by other mechanism between the ISUP and SIP networks, e.g. =
CCBS possible indicator.

Ian Elz



From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of =
"WORLEY, Dale R (Dale)" <dworley@avaya.com>

________________________________

    From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On =
Behalf Of DOLLY, MARTIN C (ATTLABS) [md3135@att.com]

    ISUP, though there is an overlap with ISDN

    If from ISDN, then there is an interworking ISDN-->ISUP-->SIP
_______________________________________________

One thing I haven't seen discussed is what is the meaning of carrying =
the ISUP release code transparently if there is more than one variety of =
ISUP, if the same release code has different meanings in different PSTN =
networks.  In a perfect world, SIP would carry not only the ISUP release =
code, but some identification of the *system* of ISUP release codes that =
it was drawn from.

Dale


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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D542585113-24062010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Hi Ian,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D542585113-24062010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>the location field for the Reason header was =
discussed in=20
the past. But it was decided not to include to many information. My=20
impression&nbsp;is that the location field is not used often for =
services or=20
other network feature. The only value would be for O&amp;M reasons where =
more=20
information is given. So your analysis in the last sentence is=20
correct.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D542585113-24062010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D542585113-24062010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>This draft is not extending the Reason header =
it is only=20
specifying the use of the Reason Header within =
responses.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D542585113-24062010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D542585113-24062010><FONT face=3DArial color=3D#0000ff =
size=3D2>Best=20
Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D542585113-24062010><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D542585113-24062010><FONT face=3DArial color=3D#0000ff =

size=3D2>Roland</FONT></SPAN></DIV><!-- Converted from text/rtf format =
-->
<P><SPAN lang=3Den-gb><FONT face=3DArial size=3D2>Mit freundlichen=20
Gr&#xFC;&#xDF;en</FONT></SPAN> <BR><SPAN lang=3Den-gb><FONT face=3DArial =
size=3D2>Roland=20
Jesske</FONT></SPAN> </P><BR>
<P><SPAN lang=3Den-gb><FONT face=3DArial color=3D#666666 =
size=3D1>Deutsche Telekom=20
Netzproduktion GmbH</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =

color=3D#808080 size=3D1>Fixed Mobile Engineering =
Deutschland</FONT></SPAN><SPAN=20
lang=3Den-gb></SPAN> <BR><SPAN lang=3Den-gb><FONT face=3DArial =
color=3D#666666=20
size=3D1>Roland Jesske</FONT></SPAN> <BR><SPAN lang=3Den-gb><FONT =
face=3DArial=20
color=3D#666666 size=3D1>Heinrich-Hertz-Stra&#xDF;e 3-7, 64295 =
Darmstadt</FONT></SPAN>=20
<BR><SPAN lang=3Den-gb><FONT face=3DArial color=3D#666666 size=3D1>+49 =
6151 628-2766=20
(Tel.)</FONT></SPAN> <BR><SPAN lang=3Den-gb><FONT face=3DArial =
color=3D#666666=20
size=3D1>+49 521 9210-3753&nbsp; (Fax)</FONT></SPAN> <BR><SPAN =
lang=3Den-gb><FONT=20
face=3DArial color=3D#666666 size=3D1>+49 171 8618-445&nbsp; =
(Mobil)</FONT></SPAN>=20
<BR><SPAN lang=3Den-gb><FONT face=3DArial color=3D#666666 =
size=3D1>E-Mail:=20
</FONT></SPAN><A href=3D"mailto:r.jesske@telekom.de"><SPAN =
lang=3Den-gb><FONT=20
face=3DArial color=3D#666666 =
size=3D1>r.jesske@telekom.de</FONT></SPAN></A><SPAN=20
lang=3Den-gb></SPAN><SPAN lang=3Dde></SPAN> <BR><SPAN =
lang=3Dde></SPAN><A=20
href=3D"http:www.telekom.com"><SPAN lang=3Dde><FONT face=3DArial =
color=3D#666666=20
size=3D1>www.telekom.com</FONT></SPAN><SPAN lang=3Dde></SPAN></A><SPAN=20
lang=3Dde></SPAN><SPAN lang=3Dde></SPAN><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>&nbsp;</FONT><FONT face=3DArial color=3D#e20074 size=3D1> =
</FONT></SPAN></P>
<P><SPAN lang=3Dde><FONT face=3DArial color=3D#e20074 size=3D1>Erleben, =
was verbindet.=20
</FONT></SPAN></P>
<P><SPAN lang=3Dde><FONT face=3DArial color=3D#666666 size=3D1>Deutsche =
Telekom=20
Netzproduktion GmbH</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =

color=3D#666666 size=3D1>Aufsichtsrat: Dr. Steffen Roehn=20
(Vorsitzender)</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
color=3D#666666=20
size=3D1>Gesch&#xE4;ftsf&#xFC;hrung: Dr. Bruno Jacobfeuerborn =
(Vorsitzender), Albert=20
Matheis, Klaus Peren</FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
color=3D#666666 size=3D1>Handelsregister: Amtsgericht Bonn HRB =
14190</FONT></SPAN>=20
<BR><SPAN lang=3Dde><FONT face=3DArial color=3D#666666 size=3D1>Sitz der =
Gesellschaft:=20
Bonn</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
color=3D#666666=20
size=3D1>USt-IdNr. DE 814645262</FONT></SPAN> </P>
<P><SPAN lang=3Dde><FONT face=3DArial color=3D#666666 =
size=3D1>Gro&#xDF;e Ver&#xE4;nderungen=20
fangen klein an</FONT> <FONT face=3DTahoma color=3D#666666 =
size=3D1>&#x2013;</FONT><FONT=20
face=3DArial color=3D#666666 size=3D1> Ressourcen schonen und nicht jede =
E-Mail=20
drucken.</FONT></SPAN> </P>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>Von:</B> Ian Elz =
[mailto:ian_elz@yahoo.co.uk]=20
<BR><B>Gesendet:</B> Donnerstag, 24. Juni 2010 15:50<BR><B>An:</B>=20
dispatch@ietf.org; WORLEY, Dale R (Dale)<BR><B>Cc:</B> DOLLY, MARTIN C=20
(ATTLABS); Tom Taylor; Jesske, Roland<BR><B>Betreff:</B> Re: [dispatch] =
ReasonIn=20
responses(draft-jesske-dispatch-reason-in-responses-02)<BR></FONT><BR></D=
IV>
<DIV></DIV>
<DIV style=3D"FONT-SIZE: 10pt; FONT-FAMILY: arial,helvetica,sans-serif">
<DIV>Dale,<BR><BR>The Q.850 cause values are standardised across the =
ISUP=20
network. As ISUP is used to interconnect the PSTN and PLMN worldwide the =
values=20
are universal.<BR><BR>The fact that the Reason header will indicate a =
protocol=20
of "Q.850" <BR><BR>The only thing that is not considered in this draft =
is that=20
there may be 'location' and 'diagnostic' information associated the =
cause=20
values.<BR><BR>The Location indicates if the release cause comes from =
teh user,=20
private networl, public network, transit network etc.<BR><BR>The =
Diagnostics=20
provides additional information as to why the realease occured and other =

information which may be cause specific.<BR><BR>This additional =
information is=20
not usually used and I don't believe that it is the intention of this =
draft to=20
expand the Reason header to carry the additional information. Any =
information=20
which is required will need to be mapped by other mechanism between the =
ISUP and=20
SIP networks, e.g. CCBS possible indicator.<BR><BR>Ian Elz<BR></DIV>
<DIV=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
arial,helvetica,sans-serif"><BR><BR>From:=20
<A href=3D"mailto:dispatch-bounces@ietf.org"=20
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A=
> [<A=20
href=3D"mailto:dispatch-bounces@ietf.org"=20
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A=
>] On=20
Behalf Of "WORLEY, Dale R (Dale)" &lt;dworley@avaya.com&gt;<BR>
<DIV style=3D"FONT-SIZE: 13px; FONT-FAMILY: =
arial,helvetica,sans-serif"><FONT=20
face=3DTahoma size=3D2>
<HR SIZE=3D1>
<B><SPAN style=3D"FONT-WEIGHT: =
bold"></SPAN></B></FONT>&nbsp;&nbsp;&nbsp; From: <A=20
href=3D"mailto:dispatch-bounces@ietf.org"=20
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A=
> [<A=20
href=3D"mailto:dispatch-bounces@ietf.org"=20
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A=
>] On=20
Behalf Of DOLLY, MARTIN C (ATTLABS) [<A href=3D"mailto:md3135@att.com"=20
ymailto=3D"mailto:md3135@att.com">md3135@att.com</A>]<BR><BR>&nbsp;&nbsp;=
&nbsp;=20
ISUP, though there is an overlap with ISDN<BR><BR>&nbsp;&nbsp;&nbsp; If =
from=20
ISDN, then there is an interworking=20
ISDN--&gt;ISUP--&gt;SIP<BR>______________________________________________=
_<BR><BR>One=20
thing I haven't seen discussed is what is the meaning of carrying the =
ISUP=20
release code transparently if there is more than one variety of ISUP, if =
the=20
same release code has different meanings in different PSTN =
networks.&nbsp; In a=20
perfect world, SIP would carry not only the ISUP release code, but some=20
identification of the *system* of ISUP release codes that it was drawn=20
from.<BR><BR>Dale</DIV></DIV></DIV><BR></BODY></HTML>

------_=_NextPart_001_01CB13A5.3ADA5F74--

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From: "Calme, James A (Jim)" <jim.calme@alcatel-lucent.com>
To: Ian Elz <ian_elz@yahoo.co.uk>, "dispatch@ietf.org" <dispatch@ietf.org>, "WORLEY, Dale R (Dale)" <dworley@avaya.com>
Date: Thu, 24 Jun 2010 09:30:32 -0500
Thread-Topic: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-responses-02)
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Subject: Re: [dispatch] ReasonIn	responses(draft-jesske-dispatch-reason-in-responses-02)
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Content-Type: text/plain; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

Ian,

It is not true that cause values are "universal". I can agree that Q.850 is=
 the starting point, but there are many countries that define their own nat=
ional extensions.

Just consider the US network as one example. The US ANSI ISUP standard T1.1=
13 defines a number of cause values in addition to Q.850. In some cases the=
 values overlap with Q.850 definitions and in other cases they are making u=
se of unused values. For the ANSI specific values the Cause Indicator codin=
g standards is set to 10 to indicate that the value is based on a national =
standard. Both Q.850 and ANSI-specific values are used within US networks.

With this is mind, where RFC 3326 defines the protocol as
protocol          =3D  "SIP" / "Q.850" / token
for US networks the value "ANSI" is introduced as a token value.

So there are networks where Reason header protocol values other than just "=
Q.850" are being used to represent ISUP cause values.

Regards,
Jim


________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Ian Elz
Sent: Thursday, June 24, 2010 8:50 AM
To: dispatch@ietf.org; WORLEY, Dale R (Dale)
Cc: DOLLY, MARTIN C (ATTLABS); R.Jesske@telekom.de
Subject: Re: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-=
responses-02)

Dale,

The Q.850 cause values are standardised across the ISUP network. As ISUP is=
 used to interconnect the PSTN and PLMN worldwide the values are universal.

The fact that the Reason header will indicate a protocol of "Q.850"

The only thing that is not considered in this draft is that there may be 'l=
ocation' and 'diagnostic' information associated the cause values.

The Location indicates if the release cause comes from teh user, private ne=
tworl, public network, transit network etc.

The Diagnostics provides additional information as to why the realease occu=
red and other information which may be cause specific.

This additional information is not usually used and I don't believe that it=
 is the intention of this draft to expand the Reason header to carry the ad=
ditional information. Any information which is required will need to be map=
ped by other mechanism between the ISUP and SIP networks, e.g. CCBS possibl=
e indicator.

Ian Elz


From: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> [dispatch=
-bounces@ietf.org<mailto:dispatch-bounces@ietf.org>] On Behalf Of "WORLEY, =
Dale R (Dale)" <dworley@avaya.com>
________________________________
    From: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> [disp=
atch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org>] On Behalf Of DOLLY=
, MARTIN C (ATTLABS) [md3135@att.com<mailto:md3135@att.com>]

    ISUP, though there is an overlap with ISDN

    If from ISDN, then there is an interworking ISDN-->ISUP-->SIP
_______________________________________________

One thing I haven't seen discussed is what is the meaning of carrying the I=
SUP release code transparently if there is more than one variety of ISUP, i=
f the same release code has different meanings in different PSTN networks. =
 In a perfect world, SIP would carry not only the ISUP release code, but so=
me identification of the *system* of ISUP release codes that it was drawn f=
rom.

Dale


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<body lang=3DEN-US link=3Dblue vlink=3Dblue>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Ian,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>It is not true that cause values are &=
#8220;universal&#8221;.
I can agree that Q.850 is the starting point, but there are many countries =
that
define their own national extensions.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Just consider the <st1:country-region
w:st=3D"on"><st1:place w:st=3D"on">US</st1:place></st1:country-region> netw=
ork as
one example. The US ANSI ISUP standard T1.113 defines a number of cause val=
ues
in addition to Q.850. In some cases the values overlap with Q.850 definitio=
ns
and in other cases they are making use of unused values. For the ANSI speci=
fic
values the Cause Indicator coding standards is set to 10 to indicate that t=
he
value is based on a national standard. Both Q.850 and ANSI-specific values =
are
used within US networks.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>With this is mind, where RFC 3326 defi=
nes
the protocol as<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 face=3D"Cour=
ier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>protocol&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
=3D&nbsp; &quot;SIP&quot; / &quot;Q.850&quot; / token<o:p></o:p></span></fo=
nt></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>for US networks the value &#8220;ANSI&=
#8221;
is introduced as a token value.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>So there are networks where Reason hea=
der protocol
values other than just &#8220;Q.850&#8221; are being used to represent ISUP
cause values.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Jim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'margin-left:.5in;text-align:=
center'><font
size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><b><font size=3D2 face=3DTa=
homa><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span>=
</font></b><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Ian Elz<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, June 24, 201=
0 8:50
AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> dispatch@ietf.org; WORLE=
Y,
Dale R (Dale)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <st1:PersonName w:st=3D"=
on">DOLLY,
 MARTIN C</st1:PersonName> (ATTLABS); R.Jesske@telekom.de<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [dispatch] Reas=
onIn responses(draft-jesske-dispatch-reason-in-responses-02)</span></font><=
o:p></o:p></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 face=3D"Time=
s New Roman"><span
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 face=3DArial=
><span
style=3D'font-size:10.0pt;font-family:Arial'>Dale,<br>
<br>
The Q.850 cause values are standardised across the ISUP network. As ISUP is
used to interconnect the PSTN and PLMN worldwide the values are universal.<=
br>
<br>
The fact that the Reason header will indicate a protocol of &quot;Q.850&quo=
t; <br>
<br>
The only thing that is not considered in this draft is that there may be
'location' and 'diagnostic' information associated the cause values.<br>
<br>
The Location indicates if the release cause comes from teh user, private
networl, public network, transit network etc.<br>
<br>
The Diagnostics provides additional information as to why the realease occu=
red
and other information which may be cause specific.<br>
<br>
This additional information is not usually used and I don't believe that it=
 is
the intention of this draft to expand the Reason header to carry the additi=
onal
information. Any information which is required will need to be mapped by ot=
her
mechanism between the ISUP and SIP networks, e.g. CCBS possible indicator.<=
br>
<br>
Ian Elz<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 face=3DArial=
><span
style=3D'font-size:10.0pt;font-family:Arial'><br>
<br>
From: <a href=3D"mailto:dispatch-bounces@ietf.org"
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a> =
[<a
href=3D"mailto:dispatch-bounces@ietf.org"
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a>]=
 On
Behalf Of &quot;WORLEY, Dale R (Dale)&quot; &lt;dworley@avaya.com&gt;<o:p><=
/o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'margin-left:.5in;text-align:=
center'><font
size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>

<hr size=3D1 width=3D"100%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 face=3DArial=
><span
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;&nbsp;&nbsp; From: <a
href=3D"mailto:dispatch-bounces@ietf.org"
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a> =
[<a
href=3D"mailto:dispatch-bounces@ietf.org"
ymailto=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a>]=
 On
Behalf Of <st1:PersonName w:st=3D"on">DOLLY, MARTIN C</st1:PersonName> (ATT=
LABS) [<a
href=3D"mailto:md3135@att.com" ymailto=3D"mailto:md3135@att.com">md3135@att=
.com</a>]<br>
<br>
&nbsp;&nbsp;&nbsp; ISUP, though there is an overlap with ISDN<br>
<br>
&nbsp;&nbsp;&nbsp; If from ISDN, then there is an interworking
ISDN--&gt;ISUP--&gt;SIP<br>
_______________________________________________<br>
<br>
One thing I haven't seen discussed is what is the meaning of carrying the I=
SUP
release code transparently if there is more than one variety of ISUP, if th=
e
same release code has different meanings in different PSTN networks.&nbsp; =
In a
perfect world, SIP would carry not only the ISUP release code, but some
identification of the *system* of ISUP release codes that it was drawn from=
.<br>
<br>
Dale<o:p></o:p></span></font></p>

</div>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 face=3D"Time=
s New Roman"><span
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

--_000_DE08556E79FD1649AEE892A8C353EF611346AC980BUSNAVSXCHMBSA_--

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Note this *may* change. 

> DISPATCH Session 1 (2 hours)
> Tuesday, Morning Session I 0900-1130



Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "ian_elz@yahoo.co.uk" <ian_elz@yahoo.co.uk>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 24 Jun 2010 18:00:12 -0400
Thread-Topic: [dispatch] Reason In responses (draft-jesske-dispatch-reason-in-responses-02)
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Subject: Re: [dispatch] Reason In responses (draft-jesske-dispatch-reason-in-responses-02)
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I think the situation can be clarified.  The various forms of REQ-1 aren't =
really requirements, as they are difficult to verify whether or not they ha=
ve been fully satisfied.  But they form very good statement of the *intenti=
on* of the requirements.  I propose changing it to a "preamble" to the requ=
irements.  REQ-2/REQ-3 aren't really distinct, but do need to be expanded w=
ith a warning regarding that "variants" may need to be tagged.  (The exact =
problems can be addressed by knowlegable people when formulating the soluti=
on.)  And we need to state that an extension mechanism must be provided.  (=
Although the Reason header already has such a mechanism, so it won't be dif=
ficult to solve that problem.)

So I get this text:


In SIP-to-PSTN gateway scenarios, it is desirable to provide the UAC
with the specific call release reason provided by the PSTN.  To
support this:

REQ:  Provide in SIP responses a way to carry PSTN call release codes,
along with indication of any context or variant identification needed
to interpret the code unambiguously.

REQ:  Provide an extensibility mechanism so that further information
about the call release can be specified.


Dale

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From: Ian Elz <ian_elz@yahoo.co.uk>
To: "Calme, James A \(Jim\)" <jim.calme@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "WORLEY, Dale R \(Dale\)" <dworley@avaya.com>
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Cc: "DOLLY, MARTIN C \(ATTLABS\)" <md3135@att.com>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
Subject: Re: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-responses-02)
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--0-1538935183-1277420691=:8793
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Jim,=0A=0A=0AI agree.=0A=0AThe protocol specifies meaning of the cause prov=
ided. I should have quoted Q.850 as an example as it is the one specificall=
y mentioned in RFC3326.=0A=0AAny other specified protocol will have its own=
 values.=0A=0AIt is up to the PSTN-SIP gateway to understand the specified =
protocol and the mapping of the cause values for that protocol.=0A=0ACheers=
=0A=0AIan=0A=0A=0A=0A=0A________________________________=0AFrom: "Calme, Ja=
mes A (Jim)" <jim.calme@alcatel-lucent.com>=0ATo: Ian Elz <ian_elz@yahoo.co=
.uk>; "dispatch@ietf.org" <dispatch@ietf.org>; "WORLEY, Dale R (Dale)" <dwo=
rley@avaya.com>=0ACc: "DOLLY, MARTIN C (ATTLABS)" <md3135@att.com>; "R.Jess=
ke@telekom.de" <R.Jesske@telekom.de>=0ASent: Thu, 24 June, 2010 15:30:32=0A=
Subject: RE: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-=
responses-02)=0A=0A =0AIan,=0A =0AIt is not true that cause values are =E2=
=80=9Cuniversal=E2=80=9D.=0AI can agree that Q.850 is the starting point, b=
ut there are many countries that=0Adefine their own national extensions.=0A=
 =0AJust consider the US network as=0Aone example. The US ANSI ISUP standar=
d T1.113 defines a number of cause values=0Ain addition to Q.850. In some c=
ases the values overlap with Q.850 definitions=0Aand in other cases they ar=
e making use of unused values. For the ANSI specific=0Avalues the Cause Ind=
icator coding standards is set to 10 to indicate that the=0Avalue is based =
on a national standard. Both Q.850 and ANSI-specific values are=0Aused with=
in US networks.=0A =0AWith this is mind, where RFC 3326 defines=0Athe proto=
col as=0Aprotocol         =0A=3D  "SIP" / "Q.850" / token=0Afor US networks=
 the value =E2=80=9CANSI=E2=80=9D=0Ais introduced as a token value.=0A =0AS=
o there are networks where Reason header protocol=0Avalues other than just =
=E2=80=9CQ.850=E2=80=9D are being used to represent ISUP=0Acause values.=0A=
 =0ARegards,=0AJim=0A =0A =0A=0A________________________________=0A =0AFrom=
:dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of =
Ian Elz=0ASent: Thursday, June 24, 2010 8:50=0AAM=0ATo: dispatch@ietf.org; =
WORLEY,=0ADale R (Dale)=0ACc: DOLLY, MARTIN C (ATTLABS); R.Jesske@telekom.d=
e=0ASubject: Re: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason=
-in-responses-02)=0A =0ADale,=0A=0AThe Q.850 cause values are standardised =
across the ISUP network. As ISUP is=0Aused to interconnect the PSTN and PLM=
N worldwide the values are universal.=0A=0AThe fact that the Reason header =
will indicate a protocol of "Q.850" =0A=0AThe only thing that is not consid=
ered in this draft is that there may be=0A'location' and 'diagnostic' infor=
mation associated the cause values.=0A=0AThe Location indicates if the rele=
ase cause comes from teh user, private=0Anetworl, public network, transit n=
etwork etc.=0A=0AThe Diagnostics provides additional information as to why =
the realease occured=0Aand other information which may be cause specific.=
=0A=0AThis additional information is not usually used and I don't believe t=
hat it is=0Athe intention of this draft to expand the Reason header to carr=
y the additional=0Ainformation. Any information which is required will need=
 to be mapped by other=0Amechanism between the ISUP and SIP networks, e.g. =
CCBS possible indicator.=0A=0AIan Elz=0A=0A=0AFrom: dispatch-bounces@ietf.o=
rg [dispatch-bounces@ietf.org] On=0ABehalf Of "WORLEY, Dale R (Dale)" <dwor=
ley@avaya.com>=0A=0A________________________________=0A =0A    From: dispat=
ch-bounces@ietf.org [dispatch-bounces@ietf.org] On=0ABehalf Of DOLLY, MARTI=
N C (ATTLABS) [md3135@att.com]=0A=0A    ISUP, though there is an overlap wi=
th ISDN=0A=0A    If from ISDN, then there is an interworking=0AISDN-->ISUP-=
->SIP=0A_______________________________________________=0A=0AOne thing I ha=
ven't seen discussed is what is the meaning of carrying the ISUP=0Arelease =
code transparently if there is more than one variety of ISUP, if the=0Asame=
 release code has different meanings in different PSTN networks.  In a=0Ape=
rfect world, SIP would carry not only the ISUP release code, but some=0Aide=
ntification of the *system* of ISUP release codes that it was drawn from.=
=0A=0ADale=0A=0A=0A      
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<html><head><style type=3D"text/css"><!-- DIV {margin:0px;} --></style></he=
ad><body><div style=3D"font-family:"arial", "helvetica", sans-serif;font-si=
ze:10pt"><DIV>Jim,<BR></DIV><DIV><BR></DIV><DIV>I agree.</DIV><DIV><BR></DI=
V><DIV>The protocol specifies meaning of the cause provided. I should have =
quoted Q.850 as an example as it is the one specifically mentioned in RFC33=
26.</DIV><DIV><BR></DIV><DIV>Any other specified protocol will have its own=
 values.</DIV><DIV><BR></DIV><DIV>It is up to the PSTN-SIP gateway to under=
stand the specified protocol and the mapping of the cause values for that p=
rotocol.</DIV><DIV><BR></DIV><DIV>Cheers</DIV><DIV><BR></DIV><DIV>Ian</DIV>=
<DIV><BR></DIV><DIV style=3D"font-family:arial, helvetica, sans-serif;font-=
size:10pt"><BR><DIV style=3D"font-family:times new roman, new york, times, =
serif;font-size:12pt"><FONT size=3D"2" face=3D"Tahoma"><HR size=3D"1"><B><S=
PAN style=3D"font-weight: bold;">From:</SPAN></B> "Calme, James A (Jim)"
 &lt;jim.calme@alcatel-lucent.com&gt;<BR><B><SPAN style=3D"font-weight: bol=
d;">To:</SPAN></B> Ian Elz &lt;ian_elz@yahoo.co.uk&gt;; "dispatch@ietf.org"=
 &lt;dispatch@ietf.org&gt;; "WORLEY, Dale R (Dale)" &lt;dworley@avaya.com&g=
t;<BR><B><SPAN style=3D"font-weight: bold;">Cc:</SPAN></B> "DOLLY, MARTIN C=
 (ATTLABS)" &lt;md3135@att.com&gt;; "R.Jesske@telekom.de" &lt;R.Jesske@tele=
kom.de&gt;<BR><B><SPAN style=3D"font-weight: bold;">Sent:</SPAN></B> Thu, 2=
4 June, 2010 15:30:32<BR><B><SPAN style=3D"font-weight: bold;">Subject:</SP=
AN></B> RE: [dispatch] ReasonIn responses(draft-jesske-dispatch-reason-in-r=
esponses-02)<BR></FONT><BR>=0A=0A=0A =0A =0A=0A =0A=0A =0A=0A =0A=0A<STYLE>=
=0A<!--=0A =0A _filtered {font-family:SimSun;panose-1:2 1 6 0 3 1 1 1 1 1;}=
=0A _filtered {font-family:Tahoma;panose-1:2 11 6 4 3 5 4 4 2 4;}=0A _filte=
red {panose-1:2 1 6 0 3 1 1 1 1 1;}=0A =0Ap.MsoNormal, li.MsoNormal, div.Ms=
oNormal=0A=09{margin:0in;margin-bottom:.0001pt;font-size:12.0pt;font-family=
:"Times New Roman";}=0Aa:link, span.MsoHyperlink=0A=09{color:blue;text-deco=
ration:underline;}=0Aa:visited, span.MsoHyperlinkFollowed=0A=09{color:blue;=
text-decoration:underline;}=0Aspan.EmailStyle17=0A=09{font-family:Arial;col=
or:navy;}=0A _filtered {margin:1.0in 1.25in 1.0in 1.25in;}=0Adiv.Section1=
=0A=09{}=0A-->=0A</STYLE>=0A=0A=0A=0A=0A=0A<DIV class=3D"Section1">=0A=0A<P=
 class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN s=
tyle=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;">Ian,</SPAN></FON=
T></P> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D=
"Arial"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;"> =
=C2=A0</SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" colo=
r=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Ar=
ial;color:navy;">It is not true that cause values are =E2=80=9Cuniversal=E2=
=80=9D.=0AI can agree that Q.850 is the starting point, but there are many =
countries that=0Adefine their own national extensions.</SPAN></FONT></P> =
=0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Arial"=
><SPAN style=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;"> =C2=A0<=
/SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=3D"na=
vy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Arial;col=
or:navy;">Just consider the=0A US network as=0Aone example. The US ANSI ISU=
P standard T1.113 defines a number of cause values=0Ain addition to Q.850. =
In some cases the values overlap with Q.850 definitions=0Aand in other case=
s they are making use of unused values. For the ANSI specific=0Avalues the =
Cause Indicator coding standards is set to 10 to indicate that the=0Avalue =
is based on a national standard. Both Q.850 and ANSI-specific values are=0A=
used within US networks.</SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FO=
NT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.=
0pt;font-family:Arial;color:navy;"> =C2=A0</SPAN></FONT></P> =0A=0A<P class=
=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN style=
=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;">With this is mind, w=
here RFC 3326 defines=0Athe protocol as</SPAN></FONT></P> =0A=0A<P class=3D=
"MsoNormal" style=3D"margin-left:.5in;"><FONT size=3D"2" face=3D"Courier Ne=
w"><SPAN style=3D'font-size:10.0pt;font-family:"Courier New";'>protocol=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=0A=3D=C2=A0 "SIP" / "Q.=
850" / token</SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2=
" color=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.0pt;font-fam=
ily:Arial;color:navy;">for US networks the value =E2=80=9CANSI=E2=80=9D=0Ai=
s introduced as a token value.</SPAN></FONT></P> =0A=0A<P class=3D"MsoNorma=
l"><FONT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-si=
ze:10.0pt;font-family:Arial;color:navy;"> =C2=A0</SPAN></FONT></P> =0A=0A<P=
 class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN s=
tyle=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;">So there are net=
works where Reason header protocol=0Avalues other than just =E2=80=9CQ.850=
=E2=80=9D are being used to represent ISUP=0Acause values.</SPAN></FONT></P=
> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Aria=
l"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;"> =C2=
=A0</SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=
=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Ari=
al;color:navy;">Regards,</SPAN></FONT></P> =0A=0A<P class=3D"MsoNormal"><FO=
NT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN style=3D"=0Afont-size:10.=
0pt;font-family:Arial;color:navy;">Jim</SPAN></FONT></P> =0A=0A<P class=3D"=
MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Arial"><SPAN style=3D"=
=0Afont-size:10.0pt;font-family:Arial;color:navy;"> =C2=A0</SPAN></FONT></P=
> =0A=0A<P class=3D"MsoNormal"><FONT size=3D"2" color=3D"navy" face=3D"Aria=
l"><SPAN style=3D"=0Afont-size:10.0pt;font-family:Arial;color:navy;"> =C2=
=A0</SPAN></FONT></P> =0A=0A<DIV>=0A=0A<DIV class=3D"MsoNormal" align=3D"ce=
nter" style=3D"margin-left:.5in;text-align:center;"><FONT size=3D"3" face=
=3D"Times New Roman"><SPAN style=3D"font-size:12.0pt;">=0A=0A<HR size=3D"2"=
 width=3D"100%" align=3D"center" tabindex=3D"-1">=0A=0A</SPAN></FONT></DIV>=
=0A=0A<P class=3D"MsoNormal" style=3D"margin-left:.5in;"><B><FONT size=3D"2=
" face=3D"Tahoma"><SPAN style=3D"font-size:10.0pt;font-family:Tahoma;font-w=
eight:bold;">From:</SPAN></FONT></B><FONT size=3D"2" face=3D"Tahoma"><SPAN =
style=3D"font-size:10.0pt;font-family:Tahoma;">=0Adispatch-bounces@ietf.org=
 [mailto:dispatch-bounces@ietf.org] <B><SPAN style=3D"font-weight:bold;">On=
 Behalf Of </SPAN></B>Ian Elz<BR>=0A<B><SPAN style=3D"font-weight:bold;">Se=
nt:</SPAN></B> Thursday, June 24, 2010 8:50=0AAM<BR>=0A<B><SPAN style=3D"fo=
nt-weight:bold;">To:</SPAN></B> dispatch@ietf.org; WORLEY,=0ADale R (Dale)<=
BR>=0A<B><SPAN style=3D"font-weight:bold;">Cc:</SPAN></B> DOLLY,=0A MARTIN =
C (ATTLABS); R.Jesske@telekom.de<BR>=0A<B><SPAN style=3D"font-weight:bold;"=
>Subject:</SPAN></B> Re: [dispatch] ReasonIn responses(draft-jesske-dispatc=
h-reason-in-responses-02)</SPAN></FONT></P> =0A=0A</DIV>=0A=0A<P class=3D"M=
soNormal" style=3D"margin-left:.5in;"><FONT size=3D"3" face=3D"Times New Ro=
man"><SPAN style=3D"font-size:12.0pt;"> =C2=A0</SPAN></FONT></P> =0A=0A<DIV=
>=0A=0A<DIV>=0A=0A<P class=3D"MsoNormal" style=3D"margin-left:.5in;"><FONT =
size=3D"2" face=3D"Arial"><SPAN style=3D"font-size:10.0pt;font-family:Arial=
;">Dale,<BR>=0A<BR>=0AThe Q.850 cause values are standardised across the IS=
UP network. As ISUP is=0Aused to interconnect the PSTN and PLMN worldwide t=
he values are universal.<BR>=0A<BR>=0AThe fact that the Reason header will =
indicate a protocol of "Q.850" <BR>=0A<BR>=0AThe only thing that is not con=
sidered in this draft is that there may be=0A'location' and 'diagnostic' in=
formation associated the cause values.<BR>=0A<BR>=0AThe Location indicates =
if the release cause comes from teh user, private=0Anetworl, public network=
, transit network etc.<BR>=0A<BR>=0AThe Diagnostics provides additional inf=
ormation as to why the realease occured=0Aand other information which may b=
e cause specific.<BR>=0A<BR>=0AThis additional information is not usually u=
sed and I don't believe that it is=0Athe intention of this draft to expand =
the Reason header to carry the additional=0Ainformation. Any information wh=
ich is required will need to be mapped by other=0Amechanism between the ISU=
P and SIP networks, e.g. CCBS possible indicator.<BR>=0A<BR>=0AIan Elz</SPA=
N></FONT></P> =0A=0A</DIV>=0A=0A<DIV>=0A=0A<P class=3D"MsoNormal" style=3D"=
margin-left:.5in;"><FONT size=3D"2" face=3D"Arial"><SPAN style=3D"font-size=
:10.0pt;font-family:Arial;"><BR>=0A<BR>=0AFrom: <A rel=3D"nofollow" ymailto=
=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank" href=3D"mailto:disp=
atch-bounces@ietf.org">dispatch-bounces@ietf.org</A> [<A rel=3D"nofollow" y=
mailto=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank" href=3D"mailt=
o:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A>] On=0ABehalf Of =
"WORLEY, Dale R (Dale)" &lt;dworley@avaya.com&gt;</SPAN></FONT></P> =0A=0A<=
DIV>=0A=0A<DIV class=3D"MsoNormal" align=3D"center" style=3D"margin-left:.5=
in;text-align:center;"><FONT size=3D"2" face=3D"Tahoma"><SPAN style=3D"font=
-size:10.0pt;font-family:Tahoma;">=0A=0A<HR size=3D"1" width=3D"100%" align=
=3D"center">=0A=0A</SPAN></FONT></DIV>=0A=0A<P class=3D"MsoNormal" style=3D=
"margin-left:.5in;"><FONT size=3D"2" face=3D"Arial"><SPAN style=3D"font-siz=
e:10.0pt;font-family:Arial;">=C2=A0=C2=A0=C2=A0 From: <A rel=3D"nofollow" y=
mailto=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank" href=3D"mailt=
o:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A> [<A rel=3D"nofol=
low" ymailto=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank" href=3D=
"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A>] On=0ABeha=
lf Of DOLLY, MARTIN C (ATTLABS) [<A rel=3D"nofollow" ymailto=3D"mailto:md31=
35@att.com" target=3D"_blank" href=3D"mailto:md3135@att.com">md3135@att.com=
</A>]<BR>=0A<BR>=0A=C2=A0=C2=A0=C2=A0 ISUP, though there is an overlap with=
 ISDN<BR>=0A<BR>=0A=C2=A0=C2=A0=C2=A0 If from ISDN, then there is an interw=
orking=0AISDN--&gt;ISUP--&gt;SIP<BR>=0A____________________________________=
___________<BR>=0A<BR>=0AOne thing I haven't seen discussed is what is the =
meaning of carrying the ISUP=0Arelease code transparently if there is more =
than one variety of ISUP, if the=0Asame release code has different meanings=
 in different PSTN networks.=C2=A0 In a=0Aperfect world, SIP would carry no=
t only the ISUP release code, but some=0Aidentification of the *system* of =
ISUP release codes that it was drawn from.<BR>=0A<BR>=0ADale</SPAN></FONT><=
/P> =0A=0A</DIV>=0A=0A</DIV>=0A=0A</DIV>=0A=0A<P class=3D"MsoNormal" style=
=3D"margin-left:.5in;"><FONT size=3D"3" face=3D"Times New Roman"><SPAN styl=
e=3D"font-size:12.0pt;"> =C2=A0</SPAN></FONT></P> =0A=0A</DIV>=0A=0A=0A=0A=
=0A</DIV></DIV>=0A=0A=0A</div><br>=0A=0A=0A=0A      </body></html>
--0-1538935183-1277420691=:8793--


From mary.ietf.barnes@gmail.com  Mon Jun 28 10:38:23 2010
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: DISPATCH <dispatch@ietf.org>
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Subject: [dispatch] VIPR - proposed charter version 3
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Hi all,

Below, please find version 3 of the VIPR proposed charter. It's been
updated to reflect ML feedback, in particular VAP has been added and
clarifications have been made with regards to impacts (i.e., none) on
existing PSTN interfaces.

Regards,
Mary
(DISPATCH WG co-chair)

============================================
VIPR Charter Proposal (Version 3)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications used
by more than a billion people daily - phone numbers and DNS rooted
address such as web servers and email addresses. The inter-domain
signaling design of SIP is primarily designed for email style addresses
yet a large percentage of SIP deployments mostly use phone numbers for
identifying users. The goal of this working group is to enable
inter-domain communications over the the Internet, using protocols such
as SIP, while still allowing people to use phone numbers to identify the
person with whom they wish to communicate.

The VIPR WG will address this problem by developing a peer to peer based
approach to finding domains that claim to be responsible for a given
phone number and validation protocols to ensure a reasonable likelihood
that a given domain actually is responsible for the phone number. In
this context, "responsible" means an administrative domain, which is at
least one of the domains, to which a PSTN call to this phone number
would be routed. Once the domain responsible for the phone number is
found, existing protocols, such as SIP, can be used for inter-domain
communications.

Some validation protocols may be based on knowledge gathered around a
PSTN call; for example, the ability to prove a call was received over
the PSTN based on start and stop times. Other validation schemes, such
as examining fingerprints or watermarking of PSTN media to show that a
domain received a particular PSTN phone call, may also be considered by
the working group. This validation will be accomplished using publicly
available open interfaces to the PSTN, so the validation can be
performed by any domain wishing to participate.  The WG will select and
standardize at least one validation scheme.

The validation mechanism requires a domain to gather and maintain
information related to PSTN calls.  This information is used by call
agents such as phones, SBCs and IP PBXs to route calls.  The WG will
define a client-server protocol between these call agents and the entity
within a domain that maintains the information.

To help mitigate SPAM issues when using SIP between domains, the WG will
define a mechanism to enable one domain to check that incoming SIP
messages are coming from a validated phone number.  A phone number is
considered validated if it is coming from a domain to which the calling
domain had previously successfully placed a PSTN call.  The working
group will define new SIP headers and option tags, as necessary, to
enable this.

The essential characteristic of VIPR is establishing authentication by
PSTN reachability when it is not possible to use a direct reference to
ENUM databases or other direct assertions of PSTN number
ownership. Elements such as public ENUM easily coexist with VIPR but no
direct interaction with ENUM will be required.  The solution set defined
by this WG will be incrementally deployable using only existing
interfaces to the PSTN.  No changes will be required to existing PSTN
capabilities, no new database access is needed nor is any new support
from PSTN service providers required.

The problem statement and some possible starting points for solutions
are further discussed in the following internet drafts which shall form
the bases of the WG documents:

draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam
draft-rosenberg-dispatch-vipr-vap

The working group will carefully coordinate with the security area, O&M
area, as well as the appropriate RAI WGs such as sipcore and p2psip.

From keith.drage@alcatel-lucent.com  Mon Jun 28 10:43:09 2010
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I have a major issue with including "which shall form the bases of the WG d=
ocuments" into the charter. To me this is confusing the charter discussion =
with the working group that might eventually result deciding what its deliv=
erables, and their contents, shall be.

regards

Keith=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Monday, June 28, 2010 6:38 PM
> To: DISPATCH
> Subject: [dispatch] VIPR - proposed charter version 3
>=20
> Hi all,
>=20
> Below, please find version 3 of the VIPR proposed charter.=20
> It's been updated to reflect ML feedback, in particular VAP=20
> has been added and clarifications have been made with regards=20
> to impacts (i.e., none) on existing PSTN interfaces.
>=20
> Regards,
> Mary
> (DISPATCH WG co-chair)
>=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> VIPR Charter Proposal (Version 3)
>=20
> WG Name:  Verification Involving PSTN Reachability (VIPR)
>=20
> There are two globally deployed address spaces for=20
> communications used by more than a billion people daily -=20
> phone numbers and DNS rooted address such as web servers and=20
> email addresses. The inter-domain signaling design of SIP is=20
> primarily designed for email style addresses yet a large=20
> percentage of SIP deployments mostly use phone numbers for=20
> identifying users. The goal of this working group is to=20
> enable inter-domain communications over the the Internet,=20
> using protocols such as SIP, while still allowing people to=20
> use phone numbers to identify the person with whom they wish=20
> to communicate.
>=20
> The VIPR WG will address this problem by developing a peer to=20
> peer based approach to finding domains that claim to be=20
> responsible for a given phone number and validation protocols=20
> to ensure a reasonable likelihood that a given domain=20
> actually is responsible for the phone number. In this=20
> context, "responsible" means an administrative domain, which=20
> is at least one of the domains, to which a PSTN call to this=20
> phone number would be routed. Once the domain responsible for=20
> the phone number is found, existing protocols, such as SIP,=20
> can be used for inter-domain communications.
>=20
> Some validation protocols may be based on knowledge gathered=20
> around a PSTN call; for example, the ability to prove a call=20
> was received over the PSTN based on start and stop times.=20
> Other validation schemes, such as examining fingerprints or=20
> watermarking of PSTN media to show that a domain received a=20
> particular PSTN phone call, may also be considered by the=20
> working group. This validation will be accomplished using=20
> publicly available open interfaces to the PSTN, so the=20
> validation can be performed by any domain wishing to=20
> participate.  The WG will select and standardize at least one=20
> validation scheme.
>=20
> The validation mechanism requires a domain to gather and=20
> maintain information related to PSTN calls.  This information=20
> is used by call agents such as phones, SBCs and IP PBXs to=20
> route calls.  The WG will define a client-server protocol=20
> between these call agents and the entity within a domain that=20
> maintains the information.
>=20
> To help mitigate SPAM issues when using SIP between domains,=20
> the WG will define a mechanism to enable one domain to check=20
> that incoming SIP messages are coming from a validated phone=20
> number.  A phone number is considered validated if it is=20
> coming from a domain to which the calling domain had=20
> previously successfully placed a PSTN call.  The working=20
> group will define new SIP headers and option tags, as=20
> necessary, to enable this.
>=20
> The essential characteristic of VIPR is establishing=20
> authentication by PSTN reachability when it is not possible=20
> to use a direct reference to ENUM databases or other direct=20
> assertions of PSTN number ownership. Elements such as public=20
> ENUM easily coexist with VIPR but no direct interaction with=20
> ENUM will be required.  The solution set defined by this WG=20
> will be incrementally deployable using only existing=20
> interfaces to the PSTN.  No changes will be required to=20
> existing PSTN capabilities, no new database access is needed=20
> nor is any new support from PSTN service providers required.
>=20
> The problem statement and some possible starting points for=20
> solutions are further discussed in the following internet=20
> drafts which shall form the bases of the WG documents:
>=20
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
> draft-rosenberg-dispatch-vipr-vap
>=20
> The working group will carefully coordinate with the security=20
> area, O&M area, as well as the appropriate RAI WGs such as=20
> sipcore and p2psip.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From mary.ietf.barnes@gmail.com  Mon Jun 28 11:00:01 2010
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Keith,

That's a valid concern and we can rewrite the deliverables as
functional descriptions as we have done for other WG charters even in
cases where documents existed (e.g., overload).  The current text is
trying to say that these documents map to deliverables and can be used
as WG documents, although as it notes these should be considered
starting points (i.e., it's understood the proposed WG has change
control), but certainly the current text is somewhat misleading.

Thanks,
Mary.

On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)
<keith.drage@alcatel-lucent.com> wrote:
> I have a major issue with including "which shall form the bases of the WG=
 documents" into the charter. To me this is confusing the charter discussio=
n with the working group that might eventually result deciding what its del=
iverables, and their contents, shall be.
>
> regards
>
> Keith
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> Sent: Monday, June 28, 2010 6:38 PM
>> To: DISPATCH
>> Subject: [dispatch] VIPR - proposed charter version 3
>>
>> Hi all,
>>
>> Below, please find version 3 of the VIPR proposed charter.
>> It's been updated to reflect ML feedback, in particular VAP
>> has been added and clarifications have been made with regards
>> to impacts (i.e., none) on existing PSTN interfaces.
>>
>> Regards,
>> Mary
>> (DISPATCH WG co-chair)
>>
>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> VIPR Charter Proposal (Version 3)
>>
>> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>>
>> There are two globally deployed address spaces for
>> communications used by more than a billion people daily -
>> phone numbers and DNS rooted address such as web servers and
>> email addresses. The inter-domain signaling design of SIP is
>> primarily designed for email style addresses yet a large
>> percentage of SIP deployments mostly use phone numbers for
>> identifying users. The goal of this working group is to
>> enable inter-domain communications over the the Internet,
>> using protocols such as SIP, while still allowing people to
>> use phone numbers to identify the person with whom they wish
>> to communicate.
>>
>> The VIPR WG will address this problem by developing a peer to
>> peer based approach to finding domains that claim to be
>> responsible for a given phone number and validation protocols
>> to ensure a reasonable likelihood that a given domain
>> actually is responsible for the phone number. In this
>> context, "responsible" means an administrative domain, which
>> is at least one of the domains, to which a PSTN call to this
>> phone number would be routed. Once the domain responsible for
>> the phone number is found, existing protocols, such as SIP,
>> can be used for inter-domain communications.
>>
>> Some validation protocols may be based on knowledge gathered
>> around a PSTN call; for example, the ability to prove a call
>> was received over the PSTN based on start and stop times.
>> Other validation schemes, such as examining fingerprints or
>> watermarking of PSTN media to show that a domain received a
>> particular PSTN phone call, may also be considered by the
>> working group. This validation will be accomplished using
>> publicly available open interfaces to the PSTN, so the
>> validation can be performed by any domain wishing to
>> participate. =A0The WG will select and standardize at least one
>> validation scheme.
>>
>> The validation mechanism requires a domain to gather and
>> maintain information related to PSTN calls. =A0This information
>> is used by call agents such as phones, SBCs and IP PBXs to
>> route calls. =A0The WG will define a client-server protocol
>> between these call agents and the entity within a domain that
>> maintains the information.
>>
>> To help mitigate SPAM issues when using SIP between domains,
>> the WG will define a mechanism to enable one domain to check
>> that incoming SIP messages are coming from a validated phone
>> number. =A0A phone number is considered validated if it is
>> coming from a domain to which the calling domain had
>> previously successfully placed a PSTN call. =A0The working
>> group will define new SIP headers and option tags, as
>> necessary, to enable this.
>>
>> The essential characteristic of VIPR is establishing
>> authentication by PSTN reachability when it is not possible
>> to use a direct reference to ENUM databases or other direct
>> assertions of PSTN number ownership. Elements such as public
>> ENUM easily coexist with VIPR but no direct interaction with
>> ENUM will be required. =A0The solution set defined by this WG
>> will be incrementally deployable using only existing
>> interfaces to the PSTN. =A0No changes will be required to
>> existing PSTN capabilities, no new database access is needed
>> nor is any new support from PSTN service providers required.
>>
>> The problem statement and some possible starting points for
>> solutions are further discussed in the following internet
>> drafts which shall form the bases of the WG documents:
>>
>> draft-rosenberg-dispatch-vipr-overview
>> draft-rosenberg-dispatch-vipr-reload-usage
>> draft-rosenberg-dispatch-vipr-pvp
>> draft-rosenberg-dispatch-vipr-sip-antispam
>> draft-rosenberg-dispatch-vipr-vap
>>
>> The working group will carefully coordinate with the security
>> area, O&M area, as well as the appropriate RAI WGs such as
>> sipcore and p2psip.
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From gonzalo.camarillo@ericsson.com  Tue Jun 29 04:03:04 2010
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Subject: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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@cisco.com>
To: iesg@ietf.org <iesg@ietf.org>
CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>
>  Robert Sparks <rjsparks@nostrum.com>
> 
> Real-time Applications and Infrastructure Area Advisor:
>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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)
> _______________________________________________
> IETF-Announce mailing list
> IETF-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf-announce


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From bruno.chatras@orange-ftgroup.com  Tue Jun 29 05:02:24 2010
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From: <bruno.chatras@orange-ftgroup.com>
To: <Gonzalo.Camarillo@ericsson.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] De la part de Gonzalo Camarillo
> Envoy=E9 : mardi 29 juin 2010 13:03
> =C0 : DISPATCH
> Objet : [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
>=20
> Hi,
>=20
> FYI: note the discussion below on the IETF general list.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> -------- 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@cisco.com>
> To: iesg@ietf.org <iesg@ietf.org>
> CC: IETF Discussion Mailing List <ietf@ietf.org>
>=20
>=20
> As far as I can tell, the WG says they wants to transfer some=20
> information to achieve cross vendor interoperability.=20
> However, what I believe the charter is actually going to do=20
> is exactly the opposite of that. When you get your head=20
> around what this charter is proposing, it is going to form a=20
> more or less opaque container for transporting proprietary=20
> information in a SIP header. It's hard to imagine how this=20
> will help interoperability.
>=20
> If we wanted to have interoperability, the charter would say=20
> what information needs to be transferred and have the WG=20
> define a way to get it between systems in an operable way.=20
> What the charter for this WG actually says they are going to=20
> do is make a special container for transfer proprietary=20
> information.  There's not even willing to say what that=20
> proprietary information is used for other than things ISDN=20
> UUI which is a  non interoperable and fairly proprietary=20
> field in ISDN.
> Furthermore they have asserted that  existing containers such=20
> as SIP-T or SIP bodies can't be used for reasons that are=20
> hard to describe. (I reject the idea that because the call=20
> might not involved the PSTN, it can't use SIP-T).
>=20
> I think the folks that want to do this should get a much=20
> clear explanation of how this results in interoperability and=20
> why exist approach such as SIP-T will not work before this is=20
> chartered.
>=20
> I do think there is a need to standardize some important call=20
> control information used in call centers and related places.=20
> However, the "we need a standard container to exchange secret=20
> information WG" is a bad idea and violates the sprit of the=20
> SIP change process not to mention the mission of the IETF.
>=20
> In summary, I'm in favor of figuring out what the problems=20
> are people hope to solve with this WG and figuring out a way=20
> to write interoperable standards to achieve that. However, I=20
> think this charter should be rejected by the IESG and sent=20
> back to the drawing board. The RAI area has things of higher=20
> priority items to work on than a SIP header for transfer=20
> proprietary data.
>=20
>=20
>=20
> On Jun 22, 2010, at 10:00 , IESG Secretary wrote:
>=20
> > A new IETF working group has been proposed in the Real-time=20
> > Applications and Infrastructure Area.  The IESG has not=20
> made any determination as yet.
> > The following draft charter was submitted, and is provided for=20
> > informational purposes only. Please send your comments to the IESG=20
> > mailing list (iesg@ietf.org) by Tuesday, June 29, 2010.
> >=20
> > Call Control UUI for SIP  (cuss)
> > --------------------------------------------------
> > Current Status: Proposed Working Group
> >=20
> > Last modified: 2010-06-21
> >=20
> > Chair(s):
> >  TBD
> >=20
> > Real-time Applications and Infrastructure Area Director(s):
> >  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>  Robert Sparks=20
> > <rjsparks@nostrum.com>
> >=20
> > Real-time Applications and Infrastructure Area Advisor:
> >  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> >=20
> > Mailing Lists: TBD
> >=20
> > Description of Working Group:
> >=20
> > The Call Control UUI for SIP (CUSS) working group is chartered to=20
> > define a Session Initiation Protocol (SIP) mechanism for=20
> transporting=20
> > call-control related user-to-user information (UUI) between User=20
> > Agents.
> >=20
> > The mechanism developed in this working group is applicable in the=20
> > following situations:
> >=20
> > 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.
> >=20
> > This mechanism is not applicable in the following situations:
> >=20
> > 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).
> >=20
> > User data of the mechanism will be clearly marked with the=20
> > application, encoding, semantics, and content type,=20
> allowing policy to=20
> > be applied by UAs.  The working group will define the=20
> information that=20
> > each application must specify to utilize the mechanism.=20
> This type of=20
> > application-specific information will be specified in=20
> standards-track=20
> > documents.
> >=20
> > One important application of this mechanism is interworking=20
> with the=20
> > ISDN User to User Information Service.  This application defined by=20
> > ITU-T Q.931 is extensively deployed in the PSTN today=20
> supporting such=20
> > applications as contact centers, call centers, and automatic call=20
> > distributors (ACDs).  A major barrier to the movement of these=20
> > applications to SIP is the lack of a standard mechanism to=20
> transport=20
> > this information in SIP.  For interworking with ISDN, minimal=20
> > information about the content of the UUI is available to=20
> the PSTN-SIP=20
> > gateways.  In this case only, the content will just=20
> indicate ISDN UUI=20
> > Service 1 interworking rather than the actual content.
> >=20
> > Call control UUI is user information conveyed between user agents=20
> > during call control operations.  As a result, the=20
> information must be=20
> > conveyed with the INVITE transaction, and must survive proxy=20
> > retargeting, redirection, and transfers.  The mechanism=20
> must utilize a=20
> > minimum of SIP extensions since it will need to be=20
> supported by many=20
> > simple SIP user agents such as PSTN gateways.  The mechanism must=20
> > interwork with the existing ISDN service but should also be=20
> extensible=20
> > for use by other applications and non-ISDN protocols.
> >=20
> > Even though interworking with the PSTN is an important requirement,=20
> > call control UUI can be exchanged between native SIP=20
> clients that do=20
> > not have any ISUP support. Therefore, existing SIP-T=20
> > encapsulation-based approaches defined in RFC3372 do not meet the=20
> > requirements to transport this type of information.
> >=20
> > Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> > considered by the working group since the UUI contents carry=20
> > information that must be conveyed during session setup at the user=20
> > agent - the information must be conveyed with the INVITE=20
> transaction.
> > The information must be passed with the session setup request=20
> > (INVITE), responses to that INVITE, or session termination requests.
> > As a result, it is not possible to use INFO in these cases.
> >=20
> > The group will produce:
> >=20
> > - A problem statement and requirements document for=20
> implementing a SIP=20
> > call control UUI mechanism
> >=20
> > - A specification of the SIP extension to best meet those=20
> requirements.
> >=20
> > Goals and Milestones
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >=20
> > Sep 10 - Problem statement and requirements document to IESG
> > (Informational)
> > Mar 11 - SIP call control UUI specification to IESG (PS)=20
> > _______________________________________________
> > IETF-Announce mailing list
> > IETF-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/ietf-announce
>=20
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From gonzalo.camarillo@ericsson.com  Tue Jun 29 05:54:32 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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Hi,

note that, at this point, these discussions should take place on the
IETF general list. If we start having discussions on different lists, it
is going to become very difficult to follow them (some people may not be
subscribed to both DISPATCH and the IETF list).

Thanks,

Gonzalo


On 29/06/2010 2:59 PM, bruno.chatras@orange-ftgroup.com wrote:
> 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@ietf.org 
>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>> To: iesg@ietf.org <iesg@ietf.org>
>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>  Robert Sparks 
>>> <rjsparks@nostrum.com>
>>>
>>> Real-time Applications and Infrastructure Area Advisor:
>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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) 
>>> _______________________________________________
>>> IETF-Announce mailing list
>>> IETF-Announce@ietf.org
>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>
>>
>> Cullen Jennings
>> For corporate legal information go to:
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>


From christer.holmberg@ericsson.com  Tue Jun 29 06:09:47 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Date: Tue, 29 Jun 2010 15:09:53 +0200
Thread-Topic: [dispatch] VIPR - proposed charter version 3
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Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi,

The second paragraph says: "The VIPR WG will address this problem..."=20

What problem?

The charter does say that the problem statement is described in the referen=
ced drafts. But, shouldn't it be described in the charter? Now it seems tha=
t we first have to adopt the drafts in order to agree what the problem is..=
.

I also agree with Keith's comment regarding the usage of "shall be based on=
 this and that draft" wording.

Regards,

Christer
=20



> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: 28. kes=E4kuuta 2010 21:00
> To: DRAGE, Keith (Keith)
> Cc: DISPATCH
> Subject: Re: [dispatch] VIPR - proposed charter version 3
>=20
> Hi Keith,
>=20
> That's a valid concern and we can rewrite the deliverables as=20
> functional descriptions as we have done for other WG charters=20
> even in cases where documents existed (e.g., overload).  The=20
> current text is trying to say that these documents map to=20
> deliverables and can be used as WG documents, although as it=20
> notes these should be considered starting points (i.e., it's=20
> understood the proposed WG has change control), but certainly=20
> the current text is somewhat misleading.
>=20
> Thanks,
> Mary.
>=20
> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)=20
> <keith.drage@alcatel-lucent.com> wrote:
> > I have a major issue with including "which shall form the=20
> bases of the WG documents" into the charter. To me this is=20
> confusing the charter discussion with the working group that=20
> might eventually result deciding what its deliverables, and=20
> their contents, shall be.
> >
> > regards
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: Monday, June 28, 2010 6:38 PM
> >> To: DISPATCH
> >> Subject: [dispatch] VIPR - proposed charter version 3
> >>
> >> Hi all,
> >>
> >> Below, please find version 3 of the VIPR proposed charter.
> >> It's been updated to reflect ML feedback, in particular=20
> VAP has been=20
> >> added and clarifications have been made with regards to impacts=20
> >> (i.e., none) on existing PSTN interfaces.
> >>
> >> Regards,
> >> Mary
> >> (DISPATCH WG co-chair)
> >>
> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >> VIPR Charter Proposal (Version 3)
> >>
> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
> >>
> >> There are two globally deployed address spaces for communications=20
> >> used by more than a billion people daily - phone numbers and DNS=20
> >> rooted address such as web servers and email addresses. The=20
> >> inter-domain signaling design of SIP is primarily designed=20
> for email=20
> >> style addresses yet a large percentage of SIP deployments=20
> mostly use=20
> >> phone numbers for identifying users. The goal of this=20
> working group=20
> >> is to enable inter-domain communications over the the=20
> Internet, using=20
> >> protocols such as SIP, while still allowing people to use phone=20
> >> numbers to identify the person with whom they wish to communicate.
> >>
> >> The VIPR WG will address this problem by developing a peer to peer=20
> >> based approach to finding domains that claim to be=20
> responsible for a=20
> >> given phone number and validation protocols to ensure a reasonable=20
> >> likelihood that a given domain actually is responsible for=20
> the phone=20
> >> number. In this context, "responsible" means an administrative=20
> >> domain, which is at least one of the domains, to which a=20
> PSTN call to=20
> >> this phone number would be routed. Once the domain responsible for=20
> >> the phone number is found, existing protocols, such as SIP, can be=20
> >> used for inter-domain communications.
> >>
> >> Some validation protocols may be based on knowledge=20
> gathered around a=20
> >> PSTN call; for example, the ability to prove a call was=20
> received over=20
> >> the PSTN based on start and stop times.
> >> Other validation schemes, such as examining fingerprints or=20
> >> watermarking of PSTN media to show that a domain received a=20
> >> particular PSTN phone call, may also be considered by the working=20
> >> group. This validation will be accomplished using publicly=20
> available=20
> >> open interfaces to the PSTN, so the validation can be performed by=20
> >> any domain wishing to participate. =A0The WG will select and=20
> >> standardize at least one validation scheme.
> >>
> >> The validation mechanism requires a domain to gather and maintain=20
> >> information related to PSTN calls. =A0This information is=20
> used by call=20
> >> agents such as phones, SBCs and IP PBXs to route calls. =A0
The WG will=20
> >> define a client-server protocol between these call agents and the=20
> >> entity within a domain that maintains the information.
> >>
> >> To help mitigate SPAM issues when using SIP between=20
> domains, the WG=20
> >> will define a mechanism to enable one domain to check that=20
> incoming=20
> >> SIP messages are coming from a validated phone number. =A0A phone=20
> >> number is considered validated if it is coming from a=20
> domain to which=20
> >> the calling domain had previously successfully placed a=20
> PSTN call. =A0
> >> The working group will define new SIP headers and option tags, as=20
> >> necessary, to enable this.
> >>
> >> The essential characteristic of VIPR is establishing=20
> authentication=20
> >> by PSTN reachability when it is not possible to use a direct=20
> >> reference to ENUM databases or other direct assertions of=20
> PSTN number=20
> >> ownership. Elements such as public ENUM easily coexist=20
> with VIPR but=20
> >> no direct interaction with ENUM will be required. =A0The=20
> solution set=20
> >> defined by this WG will be incrementally deployable using only=20
> >> existing interfaces to the PSTN. =A0No changes will be required to=20
> >> existing PSTN capabilities, no new database access is=20
> needed nor is=20
> >> any new support from PSTN service providers required.
> >>
> >> The problem statement and some possible starting points=20
> for solutions=20
> >> are further discussed in the following internet drafts which shall=20
> >> form the bases of the WG documents:
> >>
> >> draft-rosenberg-dispatch-vipr-overview
> >> draft-rosenberg-dispatch-vipr-reload-usage
> >> draft-rosenberg-dispatch-vipr-pvp
> >> draft-rosenberg-dispatch-vipr-sip-antispam
> >> draft-rosenberg-dispatch-vipr-vap
> >>
> >> The working group will carefully coordinate with the=20
> security area,=20
> >> O&M area, as well as the appropriate RAI WGs such as sipcore and=20
> >> p2psip.
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From R.Jesske@telekom.de  Tue Jun 29 06:26:49 2010
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Hi all,
For the next step I would lkie to know in which direction I should go.
Dales approach or something other like the earlier proposals?=20

So I would like to try to provide a new draft the next days.

Thank you and Best Regards

Roland


-----Urspr=FCngliche Nachricht-----
Von: WORLEY, Dale R (Dale) [mailto:dworley@avaya.com]=20
Gesendet: Freitag, 25. Juni 2010 00:00
An: Jesske, Roland; ian_elz@yahoo.co.uk; dispatch@ietf.org
Cc: md3135@att.com; tom111.taylor@bell.net
Betreff: RE: [dispatch] Reason In responses =
(draft-jesske-dispatch-reason-in-responses-02)

I think the situation can be clarified.  The various forms of REQ-1 =
aren't really requirements, as they are difficult to verify whether or =
not they have been fully satisfied.  But they form very good statement =
of the *intention* of the requirements.  I propose changing it to a =
"preamble" to the requirements.  REQ-2/REQ-3 aren't really distinct, but =
do need to be expanded with a warning regarding that "variants" may need =
to be tagged.  (The exact problems can be addressed by knowlegable =
people when formulating the solution.)  And we need to state that an =
extension mechanism must be provided.  (Although the Reason header =
already has such a mechanism, so it won't be difficult to solve that =
problem.)

So I get this text:


In SIP-to-PSTN gateway scenarios, it is desirable to provide the UAC
with the specific call release reason provided by the PSTN.  To
support this:

REQ:  Provide in SIP responses a way to carry PSTN call release codes,
along with indication of any context or variant identification needed
to interpret the code unambiguously.

REQ:  Provide an extensibility mechanism so that further information
about the call release can be specified.


Dale

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To: "DISPATCH" <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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I'd like to understand precisely what is meant by "A phone number is
considered validated if it is coming from a domain to which the calling
domain had previously successfully placed a PSTN call."
More importantly, I think the WG should discuss what the criteria for
validation should be.=20
 =20
Penn Pfautz
AT&T Access Management
+1-732-420-4962

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Mary Barnes
Sent: Monday, June 28, 2010 1:38 PM
To: DISPATCH
Subject: [dispatch] VIPR - proposed charter version 3

Hi all,

Below, please find version 3 of the VIPR proposed charter. It's been
updated to reflect ML feedback, in particular VAP has been added and
clarifications have been made with regards to impacts (i.e., none) on
existing PSTN interfaces.

Regards,
Mary
(DISPATCH WG co-chair)

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
VIPR Charter Proposal (Version 3)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications used
by more than a billion people daily - phone numbers and DNS rooted
address such as web servers and email addresses. The inter-domain
signaling design of SIP is primarily designed for email style addresses
yet a large percentage of SIP deployments mostly use phone numbers for
identifying users. The goal of this working group is to enable
inter-domain communications over the the Internet, using protocols such
as SIP, while still allowing people to use phone numbers to identify the
person with whom they wish to communicate.

The VIPR WG will address this problem by developing a peer to peer based
approach to finding domains that claim to be responsible for a given
phone number and validation protocols to ensure a reasonable likelihood
that a given domain actually is responsible for the phone number. In
this context, "responsible" means an administrative domain, which is at
least one of the domains, to which a PSTN call to this phone number
would be routed. Once the domain responsible for the phone number is
found, existing protocols, such as SIP, can be used for inter-domain
communications.

Some validation protocols may be based on knowledge gathered around a
PSTN call; for example, the ability to prove a call was received over
the PSTN based on start and stop times. Other validation schemes, such
as examining fingerprints or watermarking of PSTN media to show that a
domain received a particular PSTN phone call, may also be considered by
the working group. This validation will be accomplished using publicly
available open interfaces to the PSTN, so the validation can be
performed by any domain wishing to participate.  The WG will select and
standardize at least one validation scheme.

The validation mechanism requires a domain to gather and maintain
information related to PSTN calls.  This information is used by call
agents such as phones, SBCs and IP PBXs to route calls.  The WG will
define a client-server protocol between these call agents and the entity
within a domain that maintains the information.

To help mitigate SPAM issues when using SIP between domains, the WG will
define a mechanism to enable one domain to check that incoming SIP
messages are coming from a validated phone number.  A phone number is
considered validated if it is coming from a domain to which the calling
domain had previously successfully placed a PSTN call.  The working
group will define new SIP headers and option tags, as necessary, to
enable this.

The essential characteristic of VIPR is establishing authentication by
PSTN reachability when it is not possible to use a direct reference to
ENUM databases or other direct assertions of PSTN number
ownership. Elements such as public ENUM easily coexist with VIPR but no
direct interaction with ENUM will be required.  The solution set defined
by this WG will be incrementally deployable using only existing
interfaces to the PSTN.  No changes will be required to existing PSTN
capabilities, no new database access is needed nor is any new support
from PSTN service providers required.

The problem statement and some possible starting points for solutions
are further discussed in the following internet drafts which shall form
the bases of the WG documents:

draft-rosenberg-dispatch-vipr-overview
draft-rosenberg-dispatch-vipr-reload-usage
draft-rosenberg-dispatch-vipr-pvp
draft-rosenberg-dispatch-vipr-sip-antispam
draft-rosenberg-dispatch-vipr-vap

The working group will carefully coordinate with the security area, O&M
area, as well as the appropriate RAI WGs such as sipcore and p2psip.
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From keith.drage@alcatel-lucent.com  Tue Jun 29 08:16:15 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "bruno.chatras@orange-ftgroup.com" <bruno.chatras@orange-ftgroup.com>, "Gonzalo.Camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 29 Jun 2010 17:15:25 +0200
Thread-Topic: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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 I=
SUP protocol. When carried by ISUP the envelope is bundled up in another en=
velope 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 ca=
ll establishment. The information itself is not standardised. If you want t=
o migrate a PBX from DSS1 to SIP, then you have to take this information in=
to account. The desire is not for a WG group to standardise this existing u=
sage (which in my view would be a complete non-starter), but to allow the t=
ransfer of the existing information when migrated to a SIP environment. The=
 information transferred does not form part of SIP, and should not be stand=
ardised as part of SIP.

regards

Keith



> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> bruno.chatras@orange-ftgroup.com
> Sent: Tuesday, June 29, 2010 1:00 PM
> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.org
> Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI=20
> for SIP (cuss)
>=20
> Hum, I'm a bit surprised by the comment about SIP-T. RFC3372=20
> does state that SIP-T does not come into play when there is=20
> no PSTN involved.
>=20
> At the end of clause 2 we can read the following: "4.  IP=20
> origination - IP termination: This is a case for pure SIP.=20
> SIP-T (either encapsulation or translation of ISUP) does not=20
> come into play as there is no PSTN interworking."
>=20
> Besides, SIP-T would require encapsulating a full ISUP=20
> message (e.g. IAM) while we are interested in just one=20
> parameter of the message in the context of CUSS. This would I=20
> believe be a more severe drawback if SIP-T were used for this purpose.
>=20
> Bruno
>=20
>=20
> > -----Message d'origine-----
> > De : dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] De la part de Gonzalo Camarillo=20
> > Envoy=E9 : mardi 29 juin 2010 13:03 =C0 : DISPATCH Objet :=20
> [dispatch] Fwd:=20
> > Re: WG Review: Call Control UUI for SIP (cuss)
> >=20
> > Hi,
> >=20
> > FYI: note the discussion below on the IETF general list.
> >=20
> > Cheers,
> >=20
> > Gonzalo
> >=20
> > -------- 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@cisco.com>
> > To: iesg@ietf.org <iesg@ietf.org>
> > CC: IETF Discussion Mailing List <ietf@ietf.org>
> >=20
> >=20
> > As far as I can tell, the WG says they wants to transfer some=20
> > information to achieve cross vendor interoperability.
> > However, what I believe the charter is actually going to do=20
> is exactly=20
> > the opposite of that. When you get your head around what=20
> this charter=20
> > is proposing, it is going to form a more or less opaque=20
> container for=20
> > transporting proprietary information in a SIP header. It's hard to=20
> > imagine how this will help interoperability.
> >=20
> > If we wanted to have interoperability, the charter would say what=20
> > information needs to be transferred and have the WG define a way to=20
> > get it between systems in an operable way.
> > What the charter for this WG actually says they are going to do is=20
> > make a special container for transfer proprietary information. =20
> > There's not even willing to say what that proprietary=20
> information is=20
> > used for other than things ISDN UUI which is a  non=20
> interoperable and=20
> > fairly proprietary field in ISDN.
> > Furthermore they have asserted that  existing containers=20
> such as SIP-T=20
> > or SIP bodies can't be used for reasons that are hard to=20
> describe. (I=20
> > reject the idea that because the call might not involved=20
> the PSTN, it=20
> > can't use SIP-T).
> >=20
> > I think the folks that want to do this should get a much clear=20
> > explanation of how this results in interoperability and why exist=20
> > approach such as SIP-T will not work before this is chartered.
> >=20
> > I do think there is a need to standardize some important=20
> call control=20
> > information used in call centers and related places.
> > However, the "we need a standard container to exchange secret=20
> > information WG" is a bad idea and violates the sprit of the=20
> SIP change=20
> > process not to mention the mission of the IETF.
> >=20
> > In summary, I'm in favor of figuring out what the problems=20
> are people=20
> > hope to solve with this WG and figuring out a way to write=20
> > interoperable standards to achieve that. However, I think=20
> this charter=20
> > should be rejected by the IESG and sent back to the drawing=20
> board. The=20
> > RAI area has things of higher priority items to work on than a SIP=20
> > header for transfer proprietary data.
> >=20
> >=20
> >=20
> > On Jun 22, 2010, at 10:00 , IESG Secretary wrote:
> >=20
> > > A new IETF working group has been proposed in the Real-time=20
> > > Applications and Infrastructure Area.  The IESG has not=20
> > made any determination as yet.
> > > The following draft charter was submitted, and is provided for=20
> > > informational purposes only. Please send your comments to=20
> the IESG=20
> > > mailing list (iesg@ietf.org) by Tuesday, June 29, 2010.
> > >=20
> > > Call Control UUI for SIP  (cuss)
> > > --------------------------------------------------
> > > Current Status: Proposed Working Group
> > >=20
> > > Last modified: 2010-06-21
> > >=20
> > > Chair(s):
> > >  TBD
> > >=20
> > > Real-time Applications and Infrastructure Area Director(s):
> > >  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> =20
> Robert Sparks=20
> > > <rjsparks@nostrum.com>
> > >=20
> > > Real-time Applications and Infrastructure Area Advisor:
> > >  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > >=20
> > > Mailing Lists: TBD
> > >=20
> > > Description of Working Group:
> > >=20
> > > The Call Control UUI for SIP (CUSS) working group is chartered to=20
> > > define a Session Initiation Protocol (SIP) mechanism for=20
> > transporting=20
> > > call-control related user-to-user information (UUI) between User=20
> > > Agents.
> > >=20
> > > The mechanism developed in this working group is=20
> applicable in the=20
> > > following situations:
> > >=20
> > > 1. The information is generated and consumed by an=20
> 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=20
> significantly
> > >   changed (as discussed in Section 4 of RFC 5727).
> > > 3. Generally only the User Agent Client (UAC) and User=20
> Agent Server
> > >   (UAS) are interested in the information.
> > > 4. The information is expected to survive retargeting,=20
> redirection,
> > >   and transfers.
> > > 5. SIP elements may need to apply policy about passing=20
> and screening
> > >   the information.
> > > 6. Multi-vendor interoperability is important.
> > >=20
> > > This mechanism is not applicable in the following situations:
> > >=20
> > > 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=20
> 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).
> > >=20
> > > User data of the mechanism will be clearly marked with the=20
> > > application, encoding, semantics, and content type,=20
> > allowing policy to=20
> > > be applied by UAs.  The working group will define the=20
> > information that=20
> > > each application must specify to utilize the mechanism.=20
> > This type of=20
> > > application-specific information will be specified in=20
> > standards-track=20
> > > documents.
> > >=20
> > > One important application of this mechanism is interworking=20
> > with the=20
> > > ISDN User to User Information Service.  This application=20
> defined by=20
> > > ITU-T Q.931 is extensively deployed in the PSTN today=20
> > supporting such=20
> > > applications as contact centers, call centers, and automatic call=20
> > > distributors (ACDs).  A major barrier to the movement of these=20
> > > applications to SIP is the lack of a standard mechanism to=20
> > transport=20
> > > this information in SIP.  For interworking with ISDN, minimal=20
> > > information about the content of the UUI is available to=20
> > the PSTN-SIP=20
> > > gateways.  In this case only, the content will just=20
> > indicate ISDN UUI=20
> > > Service 1 interworking rather than the actual content.
> > >=20
> > > Call control UUI is user information conveyed between user agents=20
> > > during call control operations.  As a result, the=20
> > information must be=20
> > > conveyed with the INVITE transaction, and must survive proxy=20
> > > retargeting, redirection, and transfers.  The mechanism=20
> > must utilize a=20
> > > minimum of SIP extensions since it will need to be=20
> > supported by many=20
> > > simple SIP user agents such as PSTN gateways.  The mechanism must=20
> > > interwork with the existing ISDN service but should also be=20
> > extensible=20
> > > for use by other applications and non-ISDN protocols.
> > >=20
> > > Even though interworking with the PSTN is an important=20
> requirement,=20
> > > call control UUI can be exchanged between native SIP=20
> > clients that do=20
> > > not have any ISUP support. Therefore, existing SIP-T=20
> > > encapsulation-based approaches defined in RFC3372 do not meet the=20
> > > requirements to transport this type of information.
> > >=20
> > > Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> > > considered by the working group since the UUI contents carry=20
> > > information that must be conveyed during session setup at=20
> the user=20
> > > agent - the information must be conveyed with the INVITE=20
> > transaction.
> > > The information must be passed with the session setup request=20
> > > (INVITE), responses to that INVITE, or session=20
> termination requests.
> > > As a result, it is not possible to use INFO in these cases.
> > >=20
> > > The group will produce:
> > >=20
> > > - A problem statement and requirements document for=20
> > implementing a SIP=20
> > > call control UUI mechanism
> > >=20
> > > - A specification of the SIP extension to best meet those=20
> > requirements.
> > >=20
> > > Goals and Milestones
> > > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > >=20
> > > Sep 10 - Problem statement and requirements document to IESG
> > > (Informational)
> > > Mar 11 - SIP call control UUI specification to IESG (PS)=20
> > > _______________________________________________
> > > IETF-Announce mailing list
> > > IETF-Announce@ietf.org
> > > https://www.ietf.org/mailman/listinfo/ietf-announce
> >=20
> >=20
> > Cullen Jennings
> > For corporate legal information go to:
> > http://www.cisco.com/web/about/doing_business/legal/cri/index.html
> >=20
> >=20
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From mary.ietf.barnes@gmail.com  Tue Jun 29 09:11:25 2010
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Cc: DISPATCH <dispatch@ietf.org>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Christer,

"this problem" refers to that stated in the previous paragraph, which
is stated fairly clearly.  Would rewording the first sentence in the
second paragraph to something like the following make it more clear to
you?

" The VIPR WG will develop a peer to peer based
approach to finding domains that claim to be responsible for a given
phone number along with validation protocols to ensure a reasonable
likelihood that a given domain actually is responsible for the phone
number."

The intent is not at all that the drafts MUST be adopted in order to
agree the problem statement.  The charter should be standalone and as
I responded to Keith, we can generalize the deliverables (to match the
work items that are described in the body of the charter).  Certainly,
having such detailed documents at the outset can be misleading in
terms of the intent of the WG, however, as with any documents that
might be adopted or used as a starting point for the WG documents,
change control belongs to the WG. We have lots of examples of similar
situations (within RAI) that have resulted in a significantly
different WG document (even as early as a -00) as compared to the
individual document that was the starting point for the WG document.

Thanks,
Mary.


On Tue, Jun 29, 2010 at 8:09 AM, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
> The second paragraph says: "The VIPR WG will address this problem..."
>
> What problem?
>
> The charter does say that the problem statement is described in the refer=
enced drafts. But, shouldn't it be described in the charter? Now it seems t=
hat we first have to adopt the drafts in order to agree what the problem is=
...
>
> I also agree with Keith's comment regarding the usage of "shall be based =
on this and that draft" wording.
>
> Regards,
>
> Christer
>
>
>
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> Sent: 28. kes=E4kuuta 2010 21:00
>> To: DRAGE, Keith (Keith)
>> Cc: DISPATCH
>> Subject: Re: [dispatch] VIPR - proposed charter version 3
>>
>> Hi Keith,
>>
>> That's a valid concern and we can rewrite the deliverables as
>> functional descriptions as we have done for other WG charters
>> even in cases where documents existed (e.g., overload). =A0The
>> current text is trying to say that these documents map to
>> deliverables and can be used as WG documents, although as it
>> notes these should be considered starting points (i.e., it's
>> understood the proposed WG has change control), but certainly
>> the current text is somewhat misleading.
>>
>> Thanks,
>> Mary.
>>
>> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)
>> <keith.drage@alcatel-lucent.com> wrote:
>> > I have a major issue with including "which shall form the
>> bases of the WG documents" into the charter. To me this is
>> confusing the charter discussion with the working group that
>> might eventually result deciding what its deliverables, and
>> their contents, shall be.
>> >
>> > regards
>> >
>> > Keith
>> >
>> >> -----Original Message-----
>> >> From: dispatch-bounces@ietf.org
>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> >> Sent: Monday, June 28, 2010 6:38 PM
>> >> To: DISPATCH
>> >> Subject: [dispatch] VIPR - proposed charter version 3
>> >>
>> >> Hi all,
>> >>
>> >> Below, please find version 3 of the VIPR proposed charter.
>> >> It's been updated to reflect ML feedback, in particular
>> VAP has been
>> >> added and clarifications have been made with regards to impacts
>> >> (i.e., none) on existing PSTN interfaces.
>> >>
>> >> Regards,
>> >> Mary
>> >> (DISPATCH WG co-chair)
>> >>
>> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> >> VIPR Charter Proposal (Version 3)
>> >>
>> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>> >>
>> >> There are two globally deployed address spaces for communications
>> >> used by more than a billion people daily - phone numbers and DNS
>> >> rooted address such as web servers and email addresses. The
>> >> inter-domain signaling design of SIP is primarily designed
>> for email
>> >> style addresses yet a large percentage of SIP deployments
>> mostly use
>> >> phone numbers for identifying users. The goal of this
>> working group
>> >> is to enable inter-domain communications over the the
>> Internet, using
>> >> protocols such as SIP, while still allowing people to use phone
>> >> numbers to identify the person with whom they wish to communicate.
>> >>
>> >> The VIPR WG will address this problem by developing a peer to peer
>> >> based approach to finding domains that claim to be
>> responsible for a
>> >> given phone number and validation protocols to ensure a reasonable
>> >> likelihood that a given domain actually is responsible for
>> the phone
>> >> number. In this context, "responsible" means an administrative
>> >> domain, which is at least one of the domains, to which a
>> PSTN call to
>> >> this phone number would be routed. Once the domain responsible for
>> >> the phone number is found, existing protocols, such as SIP, can be
>> >> used for inter-domain communications.
>> >>
>> >> Some validation protocols may be based on knowledge
>> gathered around a
>> >> PSTN call; for example, the ability to prove a call was
>> received over
>> >> the PSTN based on start and stop times.
>> >> Other validation schemes, such as examining fingerprints or
>> >> watermarking of PSTN media to show that a domain received a
>> >> particular PSTN phone call, may also be considered by the working
>> >> group. This validation will be accomplished using publicly
>> available
>> >> open interfaces to the PSTN, so the validation can be performed by
>> >> any domain wishing to participate. =A0The WG will select and
>> >> standardize at least one validation scheme.
>> >>
>> >> The validation mechanism requires a domain to gather and maintain
>> >> information related to PSTN calls. =A0This information is
>> used by call
>> >> agents such as phones, SBCs and IP PBXs to route calls.
> The WG will
>> >> define a client-server protocol between these call agents and the
>> >> entity within a domain that maintains the information.
>> >>
>> >> To help mitigate SPAM issues when using SIP between
>> domains, the WG
>> >> will define a mechanism to enable one domain to check that
>> incoming
>> >> SIP messages are coming from a validated phone number. =A0A phone
>> >> number is considered validated if it is coming from a
>> domain to which
>> >> the calling domain had previously successfully placed a
>> PSTN call.
>> >> The working group will define new SIP headers and option tags, as
>> >> necessary, to enable this.
>> >>
>> >> The essential characteristic of VIPR is establishing
>> authentication
>> >> by PSTN reachability when it is not possible to use a direct
>> >> reference to ENUM databases or other direct assertions of
>> PSTN number
>> >> ownership. Elements such as public ENUM easily coexist
>> with VIPR but
>> >> no direct interaction with ENUM will be required. =A0The
>> solution set
>> >> defined by this WG will be incrementally deployable using only
>> >> existing interfaces to the PSTN. =A0No changes will be required to
>> >> existing PSTN capabilities, no new database access is
>> needed nor is
>> >> any new support from PSTN service providers required.
>> >>
>> >> The problem statement and some possible starting points
>> for solutions
>> >> are further discussed in the following internet drafts which shall
>> >> form the bases of the WG documents:
>> >>
>> >> draft-rosenberg-dispatch-vipr-overview
>> >> draft-rosenberg-dispatch-vipr-reload-usage
>> >> draft-rosenberg-dispatch-vipr-pvp
>> >> draft-rosenberg-dispatch-vipr-sip-antispam
>> >> draft-rosenberg-dispatch-vipr-vap
>> >>
>> >> The working group will carefully coordinate with the
>> security area,
>> >> O&M area, as well as the appropriate RAI WGs such as sipcore and
>> >> p2psip.
>> >> _______________________________________________
>> >> dispatch mailing list
>> >> dispatch@ietf.org
>> >> https://www.ietf.org/mailman/listinfo/dispatch
>> >>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From mary.ietf.barnes@gmail.com  Tue Jun 29 09:17:28 2010
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Penn,

Absolutely, the WG would discuss the criteria for validation - the
last sentence in the 3rd paragraph discusses this and notes:
" The WG will select and standardize at least one validation scheme."

The sentence you are asking about is in the SPAM section which is
something that happens on top of the validation - i.e., if phone
numbers associated with a specific domain have been validated, this
validation criteria is applicable to other numbers from the same
domain.

Thanks,
Mary.

On Tue, Jun 29, 2010 at 8:30 AM, PFAUTZ, PENN L (ATTCORP)
<pp3129@att.com> wrote:
> I'd like to understand precisely what is meant by "A phone number is
> considered validated if it is coming from a domain to which the calling
> domain had previously successfully placed a PSTN call."
> More importantly, I think the WG should discuss what the criteria for
> validation should be.
>
> Penn Pfautz
> AT&T Access Management
> +1-732-420-4962
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Mary Barnes
> Sent: Monday, June 28, 2010 1:38 PM
> To: DISPATCH
> Subject: [dispatch] VIPR - proposed charter version 3
>
> Hi all,
>
> Below, please find version 3 of the VIPR proposed charter. It's been
> updated to reflect ML feedback, in particular VAP has been added and
> clarifications have been made with regards to impacts (i.e., none) on
> existing PSTN interfaces.
>
> Regards,
> Mary
> (DISPATCH WG co-chair)
>
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> VIPR Charter Proposal (Version 3)
>
> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>
> There are two globally deployed address spaces for communications used
> by more than a billion people daily - phone numbers and DNS rooted
> address such as web servers and email addresses. The inter-domain
> signaling design of SIP is primarily designed for email style addresses
> yet a large percentage of SIP deployments mostly use phone numbers for
> identifying users. The goal of this working group is to enable
> inter-domain communications over the the Internet, using protocols such
> as SIP, while still allowing people to use phone numbers to identify the
> person with whom they wish to communicate.
>
> The VIPR WG will address this problem by developing a peer to peer based
> approach to finding domains that claim to be responsible for a given
> phone number and validation protocols to ensure a reasonable likelihood
> that a given domain actually is responsible for the phone number. In
> this context, "responsible" means an administrative domain, which is at
> least one of the domains, to which a PSTN call to this phone number
> would be routed. Once the domain responsible for the phone number is
> found, existing protocols, such as SIP, can be used for inter-domain
> communications.
>
> Some validation protocols may be based on knowledge gathered around a
> PSTN call; for example, the ability to prove a call was received over
> the PSTN based on start and stop times. Other validation schemes, such
> as examining fingerprints or watermarking of PSTN media to show that a
> domain received a particular PSTN phone call, may also be considered by
> the working group. This validation will be accomplished using publicly
> available open interfaces to the PSTN, so the validation can be
> performed by any domain wishing to participate. =A0The WG will select and
> standardize at least one validation scheme.
>
> The validation mechanism requires a domain to gather and maintain
> information related to PSTN calls. =A0This information is used by call
> agents such as phones, SBCs and IP PBXs to route calls. =A0The WG will
> define a client-server protocol between these call agents and the entity
> within a domain that maintains the information.
>
> To help mitigate SPAM issues when using SIP between domains, the WG will
> define a mechanism to enable one domain to check that incoming SIP
> messages are coming from a validated phone number. =A0A phone number is
> considered validated if it is coming from a domain to which the calling
> domain had previously successfully placed a PSTN call. =A0The working
> group will define new SIP headers and option tags, as necessary, to
> enable this.
>
> The essential characteristic of VIPR is establishing authentication by
> PSTN reachability when it is not possible to use a direct reference to
> ENUM databases or other direct assertions of PSTN number
> ownership. Elements such as public ENUM easily coexist with VIPR but no
> direct interaction with ENUM will be required. =A0The solution set define=
d
> by this WG will be incrementally deployable using only existing
> interfaces to the PSTN. =A0No changes will be required to existing PSTN
> capabilities, no new database access is needed nor is any new support
> from PSTN service providers required.
>
> The problem statement and some possible starting points for solutions
> are further discussed in the following internet drafts which shall form
> the bases of the WG documents:
>
> draft-rosenberg-dispatch-vipr-overview
> draft-rosenberg-dispatch-vipr-reload-usage
> draft-rosenberg-dispatch-vipr-pvp
> draft-rosenberg-dispatch-vipr-sip-antispam
> draft-rosenberg-dispatch-vipr-vap
>
> The working group will carefully coordinate with the security area, O&M
> area, as well as the appropriate RAI WGs such as sipcore and p2psip.
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From christer.holmberg@ericsson.com  Tue Jun 29 09:33:57 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Tue, 29 Jun 2010 18:33:59 +0200
Thread-Topic: [dispatch] VIPR - proposed charter version 3
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Cc: DISPATCH <dispatch@ietf.org>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Mary,=20

>"this problem" refers to that stated in the previous=20
>paragraph, which is stated fairly clearly.  Would rewording=20
>the first sentence in the second paragraph to something like=20
>the following make it more clear to you?
>=20
>" The VIPR WG will develop a peer to peer based approach to=20
>finding domains that claim to be responsible for a given=20
>phone number along with validation protocols to ensure a=20
>reasonable likelihood that a given domain actually is=20
>responsible for the phone number."

The text explains what the WG will do. But, as far as I understand, the cla=
imed PROBLEM is that there are existing mechanisms that for various reasons=
 can't be used.

Also, the charter currently DOES say that the problem statement is describe=
d in the drafts. So, I think it should be moved to the charter, so that we =
can agree on it as part of agreeing the charter.

Regards,

Christer






=20
>The intent is not at all that the drafts MUST be adopted in=20
>order to agree the problem statement.  The charter should be=20
>standalone and as I responded to Keith, we can generalize the=20
>deliverables (to match the work items that are described in=20
>the body of the charter).  Certainly, having such detailed=20
>documents at the outset can be misleading in terms of the=20
>intent of the WG, however, as with any documents that might=20
>be adopted or used as a starting point for the WG documents,=20
>change control belongs to the WG. We have lots of examples of=20
>similar situations (within RAI) that have resulted in a=20
>significantly different WG document (even as early as a -00)=20
>as compared to the individual document that was the starting=20
>point for the WG document.
>=20
> Thanks,
> Mary.
>=20
>=20
> On Tue, Jun 29, 2010 at 8:09 AM, Christer Holmberg=20
> <christer.holmberg@ericsson.com> wrote:
> >
> > Hi,
> >
> > The second paragraph says: "The VIPR WG will address this=20
> problem..."
> >
> > What problem?
> >
> > The charter does say that the problem statement is=20
> described in the referenced drafts. But, shouldn't it be=20
> described in the charter? Now it seems that we first have to=20
> adopt the drafts in order to agree what the problem is...
> >
> > I also agree with Keith's comment regarding the usage of=20
> "shall be based on this and that draft" wording.
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: 28. kes=E4kuuta 2010 21:00
> >> To: DRAGE, Keith (Keith)
> >> Cc: DISPATCH
> >> Subject: Re: [dispatch] VIPR - proposed charter version 3
> >>
> >> Hi Keith,
> >>
> >> That's a valid concern and we can rewrite the deliverables as=20
> >> functional descriptions as we have done for other WG=20
> charters even in=20
> >> cases where documents existed (e.g., overload). =A0The=20
> current text is=20
> >> trying to say that these documents map to deliverables and can be=20
> >> used as WG documents, although as it notes these should be=20
> considered=20
> >> starting points (i.e., it's understood the proposed WG has change=20
> >> control), but certainly the current text is somewhat misleading.
> >>
> >> Thanks,
> >> Mary.
> >>
> >> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)=20
> >> <keith.drage@alcatel-lucent.com> wrote:
> >> > I have a major issue with including "which shall form the
> >> bases of the WG documents" into the charter. To me this is=20
> confusing=20
> >> the charter discussion with the working group that might=20
> eventually=20
> >> result deciding what its deliverables, and their contents,=20
> shall be.
> >> >
> >> > regards
> >> >
> >> > Keith
> >> >
> >> >> -----Original Message-----
> >> >> From: dispatch-bounces@ietf.org
> >> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> >> Sent: Monday, June 28, 2010 6:38 PM
> >> >> To: DISPATCH
> >> >> Subject: [dispatch] VIPR - proposed charter version 3
> >> >>
> >> >> Hi all,
> >> >>
> >> >> Below, please find version 3 of the VIPR proposed charter.
> >> >> It's been updated to reflect ML feedback, in particular
> >> VAP has been
> >> >> added and clarifications have been made with regards to impacts=20
> >> >> (i.e., none) on existing PSTN interfaces.
> >> >>
> >> >> Regards,
> >> >> Mary
> >> >> (DISPATCH WG co-chair)
> >> >>
> >> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >> >> VIPR Charter Proposal (Version 3)
> >> >>
> >> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
> >> >>
> >> >> There are two globally deployed address spaces for=20
> communications=20
> >> >> used by more than a billion people daily - phone=20
> numbers and DNS=20
> >> >> rooted address such as web servers and email addresses. The=20
> >> >> inter-domain signaling design of SIP is primarily designed
> >> for email
> >> >> style addresses yet a large percentage of SIP deployments
> >> mostly use
> >> >> phone numbers for identifying users. The goal of this
> >> working group
> >> >> is to enable inter-domain communications over the the
> >> Internet, using
> >> >> protocols such as SIP, while still allowing people to use phone=20
> >> >> numbers to identify the person with whom they wish to=20
> communicate.
> >> >>
> >> >> The VIPR WG will address this problem by developing a=20
> peer to peer=20
> >> >> based approach to finding domains that claim to be
> >> responsible for a
> >> >> given phone number and validation protocols to ensure a=20
> reasonable=20
> >> >> likelihood that a given domain actually is responsible for
> >> the phone
> >> >> number. In this context, "responsible" means an administrative=20
> >> >> domain, which is at least one of the domains, to which a
> >> PSTN call to
> >> >> this phone number would be routed. Once the domain=20
> responsible for=20
> >> >> the phone number is found, existing protocols, such as=20
> SIP, can be=20
> >> >> used for inter-domain communications.
> >> >>
> >> >> Some validation protocols may be based on knowledge
> >> gathered around a
> >> >> PSTN call; for example, the ability to prove a call was
> >> received over
> >> >> the PSTN based on start and stop times.
> >> >> Other validation schemes, such as examining fingerprints or=20
> >> >> watermarking of PSTN media to show that a domain received a=20
> >> >> particular PSTN phone call, may also be considered by=20
> the working=20
> >> >> group. This validation will be accomplished using publicly
> >> available
> >> >> open interfaces to the PSTN, so the validation can be=20
> performed by=20
> >> >> any domain wishing to participate. =A0The WG will select and=20
> >> >> standardize at least one validation scheme.
> >> >>
> >> >> The validation mechanism requires a domain to gather=20
> and maintain=20
> >> >> information related to PSTN calls. =A0This information is
> >> used by call
> >> >> agents such as phones, SBCs and IP PBXs to route calls.
> > The WG will
> >> >> define a client-server protocol between these call=20
> agents and the=20
> >> >> entity within a domain that maintains the information.
> >> >>
> >> >> To help mitigate SPAM issues when using SIP between
> >> domains, the WG
> >> >> will define a mechanism to enable one domain to check that
> >> incoming
> >> >> SIP messages are coming from a validated phone number. =A0A phone=20
> >> >> number is considered validated if it is coming from a
> >> domain to which
> >> >> the calling domain had previously successfully placed a
> >> PSTN call.
> >> >> The working group will define new SIP headers and=20
> option tags, as=20
> >> >> necessary, to enable this.
> >> >>
> >> >> The essential characteristic of VIPR is establishing
> >> authentication
> >> >> by PSTN reachability when it is not possible to use a direct=20
> >> >> reference to ENUM databases or other direct assertions of
> >> PSTN number
> >> >> ownership. Elements such as public ENUM easily coexist
> >> with VIPR but
> >> >> no direct interaction with ENUM will be required. =A0The
> >> solution set
> >> >> defined by this WG will be incrementally deployable using only=20
> >> >> existing interfaces to the PSTN. =A0No changes will be=20
> required to=20
> >> >> existing PSTN capabilities, no new database access is
> >> needed nor is
> >> >> any new support from PSTN service providers required.
> >> >>
> >> >> The problem statement and some possible starting points
> >> for solutions
> >> >> are further discussed in the following internet drafts=20
> which shall=20
> >> >> form the bases of the WG documents:
> >> >>
> >> >> draft-rosenberg-dispatch-vipr-overview
> >> >> draft-rosenberg-dispatch-vipr-reload-usage
> >> >> draft-rosenberg-dispatch-vipr-pvp
> >> >> draft-rosenberg-dispatch-vipr-sip-antispam
> >> >> draft-rosenberg-dispatch-vipr-vap
> >> >>
> >> >> The working group will carefully coordinate with the
> >> security area,
> >> >> O&M area, as well as the appropriate RAI WGs such as=20
> sipcore and=20
> >> >> p2psip.
> >> >> _______________________________________________
> >> >> dispatch mailing list
> >> >> dispatch@ietf.org
> >> >> https://www.ietf.org/mailman/listinfo/dispatch
> >> >>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> =

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Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of 
>> bruno.chatras@orange-ftgroup.com
>> Sent: Tuesday, June 29, 2010 1:00 PM
>> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.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@ietf.org
>>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>>> To: iesg@ietf.org <iesg@ietf.org>
>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>  
>> Robert Sparks 
>>>> <rjsparks@nostrum.com>
>>>>
>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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) 
>>>> _______________________________________________
>>>> IETF-Announce mailing list
>>>> IETF-Announce@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>
>>> Cullen Jennings
>>> For corporate legal information go to:
>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>
>>>
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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Subject: [dispatch] About Final Charter for Session-ID:
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Hi,

By the charter and the original text named draft-kaplan-sip-session-id-02, 
this work is just focused on troubleshooting usage.

But I was told that *someone* want to use this session-id in Replace 
header usage, to do dialog-matching. I think this would be UPDATE of 
RFC3891.

So, I'd like to confirm that this charter and coming draft would not do 
such thing normatively.

Thanks,

Gao

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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">By the charter and the original text
named draft-kaplan-sip-session-id-02, this work is just focused on troubleshooting
usage.</font>
<br>
<br><font size=2 face="sans-serif">But I was told that *someone* want to
use this session-id in Replace header usage, to do dialog-matching. I think
this would be UPDATE of RFC3891.</font>
<br>
<br><font size=2 face="sans-serif">So, I'd like to confirm that this charter
and coming draft would not do such thing normatively.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
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From gonzalo.camarillo@ericsson.com  Tue Jun 29 23:49:22 2010
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Cc: DISPATCH <dispatch@ietf.org>, Roland Jesske <R.Jesske@telekom.de>
Subject: Re: [dispatch] WG Review: Call Control UUI for SIP (cuss)
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Hi,

please, keep the DISPATCH list in the cc: of this thread so that all
interested parties can participate in this discussion.

Thanks,

Gonzalo


On 29/06/2010 6:08 PM, Laura Liess wrote:
> Cullen,
> 
> Bruno is right.
> 
> SIP-T is used today for tunneling ISUP in SIP. In many cases  people
> don't implement SIP-T at all, because they intend to migrate the
> network to full SIP,  but still need to interwork with PSTN for a
> while. Should we require ISUP-capabilities for SIP application
> servers? We just need a way to transport a piece of data between SIP
> application servers, SIP end devices and PSTN. The solution can not be
> ISUP.
> 
>>>  However, I
>>> think this charter should be rejected by the IESG and sent
>>> back to the drawing board.
> 
> The need for UUI was recognized back in 2006,  when we had the first
> draft with this proposal see
> http://tools.ietf.org/id/draft-johnston-sipping-cc-uui-00.txt . We
> discussed the issue a number of times and people stated their need for
> this. We have now 2010.
> 
> Laura
> 
> 
> 2010/6/29  <bruno.chatras@orange-ftgroup.com>:
>> 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@ietf.org
>>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>>> To: iesg@ietf.org <iesg@ietf.org>
>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>  Robert Sparks
>>>> <rjsparks@nostrum.com>
>>>>
>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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)
>>>> _______________________________________________
>>>> IETF-Announce mailing list
>>>> IETF-Announce@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>
>>>
>>> Cullen Jennings
>>> For corporate legal information go to:
>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>
>>>
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
> 


From gonzalo.camarillo@ericsson.com  Tue Jun 29 23:50:49 2010
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, ietf@ietf.org
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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@ietf.org
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of
>>> bruno.chatras@orange-ftgroup.com
>>> Sent: Tuesday, June 29, 2010 1:00 PM
>>> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.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@ietf.org
>>>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>>>> To: iesg@ietf.org <iesg@ietf.org>
>>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>
>>> Robert Sparks
>>>>> <rjsparks@nostrum.com>
>>>>>
>>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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)
>>>>> _______________________________________________
>>>>> IETF-Announce mailing list
>>>>> IETF-Announce@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>>
>>>> Cullen Jennings
>>>> For corporate legal information go to:
>>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>


From keith.drage@alcatel-lucent.com  Wed Jun 30 03:33:07 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 30 Jun 2010 12:31:49 +0200
Thread-Topic: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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> It had been my impression that this content was standardized by ITU.
> If nobody can bother to standardize it, then it hardly seems=20
> worth working on.
>=20

So you believe that removing all the headers from SIP where nobody has both=
ered to standardise the contents is also valid?

ITU-T did not standardise the contents, only the protocol discriminator. It=
 was recognised at the time this was decided (somewhere before 1984) that s=
uch applications could not be standardised.

Keith=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Tuesday, June 29, 2010 6:24 PM
> To: DRAGE, Keith (Keith)
> Cc: bruno.chatras@orange-ftgroup.com;=20
> Gonzalo.Camarillo@ericsson.com; dispatch@ietf.org
> Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI=20
> for SIP (cuss)
>=20
>=20
>=20
> DRAGE, Keith (Keith) wrote:
> > The UUI information is NOT ISUP. It is a transparent=20
> envelope to the entire ISDN, so it is not part of an ISDN=20
> protocol and therefore not part of an ISUP protocol. When=20
> carried by ISUP the envelope is bundled up in another=20
> envelope with other information that ISUP carries transparently.
> >=20
> > So I believe, and have said repeatedly in the past, that=20
> references to SIP-T are irrelevant in this case.
> >=20
> > The problem we do have though is that are legacy usages of=20
> this information. In particular applications in PBXs carry it=20
> between themselves in ISDN call establishment. The=20
> information itself is not standardised. If you want to=20
> migrate a PBX from DSS1 to SIP, then you have to take this=20
> information into account. The desire is not for a WG group to=20
> standardise this existing usage (which in my view would be a=20
> complete non-starter), but to allow the transfer of the=20
> existing information when migrated to a SIP environment. The=20
> information transferred does not form part of SIP, and should=20
> not be standardised as part of SIP.
>=20
> How many different mechanisms do we have to construct for the=20
> purpose of tunneling stuff over SIP?
>=20
> Its especially bad if the stuff is neither standardized nor=20
> negotiated.
> It then just provides more opportunity for non-interoperability.
>=20
> It had been my impression that this content was standardized by ITU.
> If nobody can bother to standardize it, then it hardly seems=20
> worth working on.
>=20
> 	Thanks,
> 	Paul
>=20
> > regards
> >=20
> > Keith
> >=20
> >=20
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> >> bruno.chatras@orange-ftgroup.com
> >> Sent: Tuesday, June 29, 2010 1:00 PM
> >> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.org
> >> Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control=20
> UUI for SIP=20
> >> (cuss)
> >>
> >> Hum, I'm a bit surprised by the comment about SIP-T. RFC3372 does=20
> >> state that SIP-T does not come into play when there is no PSTN=20
> >> involved.
> >>
> >> At the end of clause 2 we can read the following: "4.  IP=20
> origination=20
> >> - IP termination: This is a case for pure SIP.
> >> SIP-T (either encapsulation or translation of ISUP) does not come=20
> >> into play as there is no PSTN interworking."
> >>
> >> Besides, SIP-T would require encapsulating a full ISUP=20
> message (e.g.=20
> >> IAM) while we are interested in just one parameter of the=20
> message in=20
> >> the context of CUSS. This would I believe be a more severe=20
> drawback=20
> >> if SIP-T were used for this purpose.
> >>
> >> Bruno
> >>
> >>
> >>> -----Message d'origine-----
> >>> De : dispatch-bounces@ietf.org
> >>> [mailto:dispatch-bounces@ietf.org] De la part de Gonzalo=20
> Camarillo=20
> >>> Envoy=E9 : mardi 29 juin 2010 13:03 =C0 : DISPATCH Objet :
> >> [dispatch] Fwd:=20
> >>> 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@cisco.com>
> >>> To: iesg@ietf.org <iesg@ietf.org>
> >>> CC: IETF Discussion Mailing List <ietf@ietf.org>
> >>>
> >>>
> >>> As far as I can tell, the WG says they wants to transfer some=20
> >>> 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.=20
> It's hard to=20
> >>> imagine how this will help interoperability.
> >>>
> >>> If we wanted to have interoperability, the charter would say what=20
> >>> information needs to be transferred and have the WG=20
> define a way to=20
> >>> get it between systems in an operable way.
> >>> What the charter for this WG actually says they are going=20
> to do is=20
> >>> 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=20
> >>> explanation of how this results in interoperability and why exist=20
> >>> 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=20
> >>> 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=20
> >>> 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=20
> than a SIP=20
> >>> 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=20
> >>>> Applications and Infrastructure Area.  The IESG has not
> >>> made any determination as yet.
> >>>> The following draft charter was submitted, and is provided for=20
> >>>> informational purposes only. Please send your comments to
> >> the IESG
> >>>> mailing list (iesg@ietf.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.Camarillo@ericsson.com>
> >> Robert Sparks
> >>>> <rjsparks@nostrum.com>
> >>>>
> >>>> Real-time Applications and Infrastructure Area Advisor:
> >>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> >>>>
> >>>> Mailing Lists: TBD
> >>>>
> >>>> Description of Working Group:
> >>>>
> >>>> The Call Control UUI for SIP (CUSS) working group is=20
> chartered to=20
> >>>> define a Session Initiation Protocol (SIP) mechanism for
> >>> transporting
> >>>> call-control related user-to-user information (UUI) between User=20
> >>>> 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=20
> information
> >>>>   being transported (e.g., geolocation or emergency-related
> >>>>   information).
> >>>>
> >>>> User data of the mechanism will be clearly marked with the=20
> >>>> 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.=20
> >>> 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=20
> automatic call=20
> >>>> distributors (ACDs).  A major barrier to the movement of these=20
> >>>> applications to SIP is the lack of a standard mechanism to
> >>> transport
> >>>> this information in SIP.  For interworking with ISDN, minimal=20
> >>>> 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=20
> user agents=20
> >>>> during call control operations.  As a result, the
> >>> information must be
> >>>> conveyed with the INVITE transaction, and must survive proxy=20
> >>>> 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=20
> mechanism must=20
> >>>> 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=20
> >>>> encapsulation-based approaches defined in RFC3372 do not=20
> meet the=20
> >>>> requirements to transport this type of information.
> >>>>
> >>>> Mechanisms based on the SIP INFO method, RFC2976, will not be=20
> >>>> considered by the working group since the UUI contents carry=20
> >>>> 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=20
> >>>> (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
> >>>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>>>
> >>>> Sep 10 - Problem statement and requirements document to IESG
> >>>> (Informational)
> >>>> Mar 11 - SIP call control UUI specification to IESG (PS)=20
> >>>> _______________________________________________
> >>>> IETF-Announce mailing list
> >>>> IETF-Announce@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/ietf-announce
> >>>
> >>> Cullen Jennings
> >>> For corporate legal information go to:
> >>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> dispatch mailing list
> >>> dispatch@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dispatch
> >>>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> =

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Subject: Re: [dispatch] VIPR - proposed charter version 3
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> The VIPR WG will address this problem by developing a peer to=20
> peer based approach to finding domains that claim to be=20
> responsible for a given phone number and validation protocols=20
> to ensure a reasonable likelihood that a given domain=20
> actually is responsible for the phone number.=20

Hi,

Clarification question. What exactly means 'peer to peer based approach'
and what kind of approaches are excluded by having this in the charter.
Does 'approach' mean solution? If so why does a specific type of
solution need to be agreed in the charter, while all we have at hand at
this point are individual contribution I-Ds that describe the 'problem
statement and some possible starting points for solutions'?

Thanks and Regards,

Dan


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Monday, June 28, 2010 8:38 PM
> To: DISPATCH
> Subject: [dispatch] VIPR - proposed charter version 3
>=20

From dromasca@avaya.com  Wed Jun 30 03:48:13 2010
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Thread-Topic: [dispatch] VIPR - proposed charter version 3
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References: <AANLkTintQWiM1BNi1Lz11i4AEUm4vnpFhHNRPRMs6ctG@mail.gmail.com><EDC0A1AE77C57744B664A310A0B23AE213F8C43A@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <AANLkTiknOb92nDGrm9sjD5LLh9mvkVqSS75SrzyGrniX@mail.gmail.com>
From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Mary Barnes" <mary.ietf.barnes@gmail.com>, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Cc: DISPATCH <dispatch@ietf.org>, IETF-Discussion list <ietf@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Mary,

I also think that listing the deliverables should be independent from =
mentioning the existing initial contributions. The existing =
contributions could be listed as well, but they should not preclude =
other contributions on the same items after the WG is formed.=20

Regards,

Dan


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Monday, June 28, 2010 9:00 PM
> To: DRAGE, Keith (Keith)
> Cc: DISPATCH
> Subject: Re: [dispatch] VIPR - proposed charter version 3
>=20
> Hi Keith,
>=20
> That's a valid concern and we can rewrite the deliverables as=20
> functional descriptions as we have done for other WG charters=20
> even in cases where documents existed (e.g., overload).  The=20
> current text is trying to say that these documents map to=20
> deliverables and can be used as WG documents, although as it=20
> notes these should be considered starting points (i.e., it's=20
> understood the proposed WG has change control), but certainly=20
> the current text is somewhat misleading.
>=20
> Thanks,
> Mary.
>=20
> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)=20
> <keith.drage@alcatel-lucent.com> wrote:
> > I have a major issue with including "which shall form the=20
> bases of the WG documents" into the charter. To me this is=20
> confusing the charter discussion with the working group that=20
> might eventually result deciding what its deliverables, and=20
> their contents, shall be.
> >
> > regards
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: Monday, June 28, 2010 6:38 PM
> >> To: DISPATCH
> >> Subject: [dispatch] VIPR - proposed charter version 3
> >>
> >> Hi all,
> >>
> >> Below, please find version 3 of the VIPR proposed charter.
> >> It's been updated to reflect ML feedback, in particular=20
> VAP has been=20
> >> added and clarifications have been made with regards to impacts=20
> >> (i.e., none) on existing PSTN interfaces.
> >>
> >> Regards,
> >> Mary
> >> (DISPATCH WG co-chair)
> >>
> >> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >> VIPR Charter Proposal (Version 3)
> >>
> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
> >>
> >> There are two globally deployed address spaces for communications=20
> >> used by more than a billion people daily - phone numbers and DNS=20
> >> rooted address such as web servers and email addresses. The=20
> >> inter-domain signaling design of SIP is primarily designed=20
> for email=20
> >> style addresses yet a large percentage of SIP deployments=20
> mostly use=20
> >> phone numbers for identifying users. The goal of this=20
> working group=20
> >> is to enable inter-domain communications over the the=20
> Internet, using=20
> >> protocols such as SIP, while still allowing people to use phone=20
> >> numbers to identify the person with whom they wish to communicate.
> >>
> >> The VIPR WG will address this problem by developing a peer to peer=20
> >> based approach to finding domains that claim to be=20
> responsible for a=20
> >> given phone number and validation protocols to ensure a reasonable=20
> >> likelihood that a given domain actually is responsible for=20
> the phone=20
> >> number. In this context, "responsible" means an administrative=20
> >> domain, which is at least one of the domains, to which a=20
> PSTN call to=20
> >> this phone number would be routed. Once the domain responsible for=20
> >> the phone number is found, existing protocols, such as SIP, can be=20
> >> used for inter-domain communications.
> >>
> >> Some validation protocols may be based on knowledge=20
> gathered around a=20
> >> PSTN call; for example, the ability to prove a call was=20
> received over=20
> >> the PSTN based on start and stop times.
> >> Other validation schemes, such as examining fingerprints or=20
> >> watermarking of PSTN media to show that a domain received a=20
> >> particular PSTN phone call, may also be considered by the working=20
> >> group. This validation will be accomplished using publicly=20
> available=20
> >> open interfaces to the PSTN, so the validation can be performed by=20
> >> any domain wishing to participate. =A0The WG will select and=20
> >> standardize at least one validation scheme.
> >>
> >> The validation mechanism requires a domain to gather and maintain=20
> >> information related to PSTN calls. =A0This information is=20
> used by call=20
> >> agents such as phones, SBCs and IP PBXs to route calls. =A0
> The WG will=20
> >> define a client-server protocol between these call agents and the=20
> >> entity within a domain that maintains the information.
> >>
> >> To help mitigate SPAM issues when using SIP between=20
> domains, the WG=20
> >> will define a mechanism to enable one domain to check that=20
> incoming=20
> >> SIP messages are coming from a validated phone number. =A0A phone=20
> >> number is considered validated if it is coming from a=20
> domain to which=20
> >> the calling domain had previously successfully placed a=20
> PSTN call. =A0
> >> The working group will define new SIP headers and option tags, as=20
> >> necessary, to enable this.
> >>
> >> The essential characteristic of VIPR is establishing=20
> authentication=20
> >> by PSTN reachability when it is not possible to use a direct=20
> >> reference to ENUM databases or other direct assertions of=20
> PSTN number=20
> >> ownership. Elements such as public ENUM easily coexist=20
> with VIPR but=20
> >> no direct interaction with ENUM will be required. =A0The=20
> solution set=20
> >> defined by this WG will be incrementally deployable using only=20
> >> existing interfaces to the PSTN. =A0No changes will be required to=20
> >> existing PSTN capabilities, no new database access is=20
> needed nor is=20
> >> any new support from PSTN service providers required.
> >>
> >> The problem statement and some possible starting points=20
> for solutions=20
> >> are further discussed in the following internet drafts which shall=20
> >> form the bases of the WG documents:
> >>
> >> draft-rosenberg-dispatch-vipr-overview
> >> draft-rosenberg-dispatch-vipr-reload-usage
> >> draft-rosenberg-dispatch-vipr-pvp
> >> draft-rosenberg-dispatch-vipr-sip-antispam
> >> draft-rosenberg-dispatch-vipr-vap
> >>
> >> The working group will carefully coordinate with the=20
> security area,=20
> >> O&M area, as well as the appropriate RAI WGs such as sipcore and=20
> >> p2psip.
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From keith.drage@alcatel-lucent.com  Wed Jun 30 04:57:40 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Date: Wed, 30 Jun 2010 13:57:32 +0200
Thread-Topic: [dispatch] VIPR - proposed charter version 3
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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It is not clear whether you are proposing a rewrite that removes the names =
of the deliverables, and therefore removes the problem that way, or not. Wi=
thout seeing words, I cannot comment on the specifics of that.

In any case, the minimum change to me is to remove the words indicated.

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Monday, June 28, 2010 7:00 PM
> To: DRAGE, Keith (Keith)
> Cc: DISPATCH
> Subject: Re: [dispatch] VIPR - proposed charter version 3
>=20
> Hi Keith,
>=20
> That's a valid concern and we can rewrite the deliverables as=20
> functional descriptions as we have done for other WG charters=20
> even in cases where documents existed (e.g., overload).  The=20
> current text is trying to say that these documents map to=20
> deliverables and can be used as WG documents, although as it=20
> notes these should be considered starting points (i.e., it's=20
> understood the proposed WG has change control), but certainly=20
> the current text is somewhat misleading.
>=20
> Thanks,
> Mary.
>=20
> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)=20
> <keith.drage@alcatel-lucent.com> wrote:
> > I have a major issue with including "which shall form the=20
> bases of the WG documents" into the charter. To me this is=20
> confusing the charter discussion with the working group that=20
> might eventually result deciding what its deliverables, and=20
> their contents, shall be.
> >
> > regards
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: Monday, June 28, 2010 6:38 PM
> >> To: DISPATCH
> >> Subject: [dispatch] VIPR - proposed charter version 3
> >>
> >> Hi all,
> >>
> >> Below, please find version 3 of the VIPR proposed charter.
> >> It's been updated to reflect ML feedback, in particular=20
> VAP has been=20
> >> added and clarifications have been made with regards to impacts=20
> >> (i.e., none) on existing PSTN interfaces.
> >>
> >> Regards,
> >> Mary
> >> (DISPATCH WG co-chair)
> >>
> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >> VIPR Charter Proposal (Version 3)
> >>
> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
> >>
> >> There are two globally deployed address spaces for communications=20
> >> used by more than a billion people daily - phone numbers and DNS=20
> >> rooted address such as web servers and email addresses. The=20
> >> inter-domain signaling design of SIP is primarily designed=20
> for email=20
> >> style addresses yet a large percentage of SIP deployments=20
> mostly use=20
> >> phone numbers for identifying users. The goal of this=20
> working group=20
> >> is to enable inter-domain communications over the the=20
> Internet, using=20
> >> protocols such as SIP, while still allowing people to use phone=20
> >> numbers to identify the person with whom they wish to communicate.
> >>
> >> The VIPR WG will address this problem by developing a peer to peer=20
> >> based approach to finding domains that claim to be=20
> responsible for a=20
> >> given phone number and validation protocols to ensure a reasonable=20
> >> likelihood that a given domain actually is responsible for=20
> the phone=20
> >> number. In this context, "responsible" means an administrative=20
> >> domain, which is at least one of the domains, to which a=20
> PSTN call to=20
> >> this phone number would be routed. Once the domain responsible for=20
> >> the phone number is found, existing protocols, such as SIP, can be=20
> >> used for inter-domain communications.
> >>
> >> Some validation protocols may be based on knowledge=20
> gathered around a=20
> >> PSTN call; for example, the ability to prove a call was=20
> received over=20
> >> the PSTN based on start and stop times.
> >> Other validation schemes, such as examining fingerprints or=20
> >> watermarking of PSTN media to show that a domain received a=20
> >> particular PSTN phone call, may also be considered by the working=20
> >> group. This validation will be accomplished using publicly=20
> available=20
> >> open interfaces to the PSTN, so the validation can be performed by=20
> >> any domain wishing to participate. =A0The WG will select and=20
> >> standardize at least one validation scheme.
> >>
> >> The validation mechanism requires a domain to gather and maintain=20
> >> information related to PSTN calls. =A0This information is=20
> used by call=20
> >> agents such as phones, SBCs and IP PBXs to route calls. =A0
> The WG will=20
> >> define a client-server protocol between these call agents and the=20
> >> entity within a domain that maintains the information.
> >>
> >> To help mitigate SPAM issues when using SIP between=20
> domains, the WG=20
> >> will define a mechanism to enable one domain to check that=20
> incoming=20
> >> SIP messages are coming from a validated phone number. =A0A phone=20
> >> number is considered validated if it is coming from a=20
> domain to which=20
> >> the calling domain had previously successfully placed a=20
> PSTN call. =A0
> >> The working group will define new SIP headers and option tags, as=20
> >> necessary, to enable this.
> >>
> >> The essential characteristic of VIPR is establishing=20
> authentication=20
> >> by PSTN reachability when it is not possible to use a direct=20
> >> reference to ENUM databases or other direct assertions of=20
> PSTN number=20
> >> ownership. Elements such as public ENUM easily coexist=20
> with VIPR but=20
> >> no direct interaction with ENUM will be required. =A0The=20
> solution set=20
> >> defined by this WG will be incrementally deployable using only=20
> >> existing interfaces to the PSTN. =A0No changes will be required to=20
> >> existing PSTN capabilities, no new database access is=20
> needed nor is=20
> >> any new support from PSTN service providers required.
> >>
> >> The problem statement and some possible starting points=20
> for solutions=20
> >> are further discussed in the following internet drafts which shall=20
> >> form the bases of the WG documents:
> >>
> >> draft-rosenberg-dispatch-vipr-overview
> >> draft-rosenberg-dispatch-vipr-reload-usage
> >> draft-rosenberg-dispatch-vipr-pvp
> >> draft-rosenberg-dispatch-vipr-sip-antispam
> >> draft-rosenberg-dispatch-vipr-vap
> >>
> >> The working group will carefully coordinate with the=20
> security area,=20
> >> O&M area, as well as the appropriate RAI WGs such as sipcore and=20
> >> p2psip.
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From gonzalo.camarillo@ericsson.com  Wed Jun 30 04:59:29 2010
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Subject: [dispatch] Community reviews
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Hi,

as you know, the IETF processes include community reviews of drafts, WG
charters, etc. These reviews are typically announced on the IETF
announce list. Discussions are supposed to take place on both the IETF
general discussion list and the WG list.

It is important that messages are sent to both lists because there are
reviewers that are not subscribed to the WG list (e.g., ADs from other
areas) and WG contributors that are not subscribed (and do not want to
be for a number of reasons) to the IETF general discussion list.

So far, I have been taking care of forwarding emails that are only sent
to one of the lists, but, as you can imagine, it would be far better if
people took care of this themselves when sending their emails.

The thing is that IETF LCs on drafts are generally automatically sent to
both lists. However, the latest requests to review charter proposals
have only been sent to the IETF general discussion list. We will try and
make sure that future requests to review charter proposals coming from
DISPATCH are also sent to the DISPATCH list... in the meantime, please
to as suggested above.

Thanks,

Gonzalo


From mary.ietf.barnes@gmail.com  Wed Jun 30 08:23:58 2010
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Date: Wed, 30 Jun 2010 10:24:01 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
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Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Dan,

The term peer to peer is intended to exclude mechanisms that would use
a central repository for the information:  This was discussed in an
earlier thread:
http://www.ietf.org/mail-archive/web/dispatch/current/msg02027.html

In one sense it is a solution, however, in another sense it is reusing
SIP related functionality defined in RAI and thus is in a similar vein
as specifying the use of SIP in a charter.

Thanks,
Mary.

On Wed, Jun 30, 2010 at 5:42 AM, Romascanu, Dan (Dan)
<dromasca@avaya.com> wrote:
>> The VIPR WG will address this problem by developing a peer to
>> peer based approach to finding domains that claim to be
>> responsible for a given phone number and validation protocols
>> to ensure a reasonable likelihood that a given domain
>> actually is responsible for the phone number.
>
> Hi,
>
> Clarification question. What exactly means 'peer to peer based approach'
> and what kind of approaches are excluded by having this in the charter.
> Does 'approach' mean solution? If so why does a specific type of
> solution need to be agreed in the charter, while all we have at hand at
> this point are individual contribution I-Ds that describe the 'problem
> statement and some possible starting points for solutions'?
>
> Thanks and Regards,
>
> Dan
>
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> Sent: Monday, June 28, 2010 8:38 PM
>> To: DISPATCH
>> Subject: [dispatch] VIPR - proposed charter version 3
>>
>

From mary.ietf.barnes@gmail.com  Wed Jun 30 08:24:46 2010
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Cc: DISPATCH <dispatch@ietf.org>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, IETF-Discussion list <ietf@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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Agreed - I will make that change in the version 4 of the charter.

Thanks,
Mary.

2010/6/30 Romascanu, Dan (Dan) <dromasca@avaya.com>:
> Hi Mary,
>
> I also think that listing the deliverables should be independent from men=
tioning the existing initial contributions. The existing contributions coul=
d be listed as well, but they should not preclude other contributions on th=
e same items after the WG is formed.
>
> Regards,
>
> Dan
>
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> Sent: Monday, June 28, 2010 9:00 PM
>> To: DRAGE, Keith (Keith)
>> Cc: DISPATCH
>> Subject: Re: [dispatch] VIPR - proposed charter version 3
>>
>> Hi Keith,
>>
>> That's a valid concern and we can rewrite the deliverables as
>> functional descriptions as we have done for other WG charters
>> even in cases where documents existed (e.g., overload). =A0The
>> current text is trying to say that these documents map to
>> deliverables and can be used as WG documents, although as it
>> notes these should be considered starting points (i.e., it's
>> understood the proposed WG has change control), but certainly
>> the current text is somewhat misleading.
>>
>> Thanks,
>> Mary.
>>
>> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)
>> <keith.drage@alcatel-lucent.com> wrote:
>> > I have a major issue with including "which shall form the
>> bases of the WG documents" into the charter. To me this is
>> confusing the charter discussion with the working group that
>> might eventually result deciding what its deliverables, and
>> their contents, shall be.
>> >
>> > regards
>> >
>> > Keith
>> >
>> >> -----Original Message-----
>> >> From: dispatch-bounces@ietf.org
>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> >> Sent: Monday, June 28, 2010 6:38 PM
>> >> To: DISPATCH
>> >> Subject: [dispatch] VIPR - proposed charter version 3
>> >>
>> >> Hi all,
>> >>
>> >> Below, please find version 3 of the VIPR proposed charter.
>> >> It's been updated to reflect ML feedback, in particular
>> VAP has been
>> >> added and clarifications have been made with regards to impacts
>> >> (i.e., none) on existing PSTN interfaces.
>> >>
>> >> Regards,
>> >> Mary
>> >> (DISPATCH WG co-chair)
>> >>
>> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> >> VIPR Charter Proposal (Version 3)
>> >>
>> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>> >>
>> >> There are two globally deployed address spaces for communications
>> >> used by more than a billion people daily - phone numbers and DNS
>> >> rooted address such as web servers and email addresses. The
>> >> inter-domain signaling design of SIP is primarily designed
>> for email
>> >> style addresses yet a large percentage of SIP deployments
>> mostly use
>> >> phone numbers for identifying users. The goal of this
>> working group
>> >> is to enable inter-domain communications over the the
>> Internet, using
>> >> protocols such as SIP, while still allowing people to use phone
>> >> numbers to identify the person with whom they wish to communicate.
>> >>
>> >> The VIPR WG will address this problem by developing a peer to peer
>> >> based approach to finding domains that claim to be
>> responsible for a
>> >> given phone number and validation protocols to ensure a reasonable
>> >> likelihood that a given domain actually is responsible for
>> the phone
>> >> number. In this context, "responsible" means an administrative
>> >> domain, which is at least one of the domains, to which a
>> PSTN call to
>> >> this phone number would be routed. Once the domain responsible for
>> >> the phone number is found, existing protocols, such as SIP, can be
>> >> used for inter-domain communications.
>> >>
>> >> Some validation protocols may be based on knowledge
>> gathered around a
>> >> PSTN call; for example, the ability to prove a call was
>> received over
>> >> the PSTN based on start and stop times.
>> >> Other validation schemes, such as examining fingerprints or
>> >> watermarking of PSTN media to show that a domain received a
>> >> particular PSTN phone call, may also be considered by the working
>> >> group. This validation will be accomplished using publicly
>> available
>> >> open interfaces to the PSTN, so the validation can be performed by
>> >> any domain wishing to participate. =A0The WG will select and
>> >> standardize at least one validation scheme.
>> >>
>> >> The validation mechanism requires a domain to gather and maintain
>> >> information related to PSTN calls. =A0This information is
>> used by call
>> >> agents such as phones, SBCs and IP PBXs to route calls.
>> The WG will
>> >> define a client-server protocol between these call agents and the
>> >> entity within a domain that maintains the information.
>> >>
>> >> To help mitigate SPAM issues when using SIP between
>> domains, the WG
>> >> will define a mechanism to enable one domain to check that
>> incoming
>> >> SIP messages are coming from a validated phone number. =A0A phone
>> >> number is considered validated if it is coming from a
>> domain to which
>> >> the calling domain had previously successfully placed a
>> PSTN call.
>> >> The working group will define new SIP headers and option tags, as
>> >> necessary, to enable this.
>> >>
>> >> The essential characteristic of VIPR is establishing
>> authentication
>> >> by PSTN reachability when it is not possible to use a direct
>> >> reference to ENUM databases or other direct assertions of
>> PSTN number
>> >> ownership. Elements such as public ENUM easily coexist
>> with VIPR but
>> >> no direct interaction with ENUM will be required. =A0The
>> solution set
>> >> defined by this WG will be incrementally deployable using only
>> >> existing interfaces to the PSTN. =A0No changes will be required to
>> >> existing PSTN capabilities, no new database access is
>> needed nor is
>> >> any new support from PSTN service providers required.
>> >>
>> >> The problem statement and some possible starting points
>> for solutions
>> >> are further discussed in the following internet drafts which shall
>> >> form the bases of the WG documents:
>> >>
>> >> draft-rosenberg-dispatch-vipr-overview
>> >> draft-rosenberg-dispatch-vipr-reload-usage
>> >> draft-rosenberg-dispatch-vipr-pvp
>> >> draft-rosenberg-dispatch-vipr-sip-antispam
>> >> draft-rosenberg-dispatch-vipr-vap
>> >>
>> >> The working group will carefully coordinate with the
>> security area,
>> >> O&M area, as well as the appropriate RAI WGs such as sipcore and
>> >> p2psip.
>> >> _______________________________________________
>> >> dispatch mailing list
>> >> dispatch@ietf.org
>> >> https://www.ietf.org/mailman/listinfo/dispatch
>> >>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
>

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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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I will remove both the list of draft names and the paragraph that
introduces those and replace it with text describing deliverables as
is typically done for charters.  At this point, I think everyone is
well aware of the related documents.

Thanks,
Mary.

On Wed, Jun 30, 2010 at 6:57 AM, DRAGE, Keith (Keith)
<keith.drage@alcatel-lucent.com> wrote:
> It is not clear whether you are proposing a rewrite that removes the name=
s of the deliverables, and therefore removes the problem that way, or not. =
Without seeing words, I cannot comment on the specifics of that.
>
> In any case, the minimum change to me is to remove the words indicated.
>
> regards
>
> Keith
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> Sent: Monday, June 28, 2010 7:00 PM
>> To: DRAGE, Keith (Keith)
>> Cc: DISPATCH
>> Subject: Re: [dispatch] VIPR - proposed charter version 3
>>
>> Hi Keith,
>>
>> That's a valid concern and we can rewrite the deliverables as
>> functional descriptions as we have done for other WG charters
>> even in cases where documents existed (e.g., overload). =A0The
>> current text is trying to say that these documents map to
>> deliverables and can be used as WG documents, although as it
>> notes these should be considered starting points (i.e., it's
>> understood the proposed WG has change control), but certainly
>> the current text is somewhat misleading.
>>
>> Thanks,
>> Mary.
>>
>> On Mon, Jun 28, 2010 at 12:43 PM, DRAGE, Keith (Keith)
>> <keith.drage@alcatel-lucent.com> wrote:
>> > I have a major issue with including "which shall form the
>> bases of the WG documents" into the charter. To me this is
>> confusing the charter discussion with the working group that
>> might eventually result deciding what its deliverables, and
>> their contents, shall be.
>> >
>> > regards
>> >
>> > Keith
>> >
>> >> -----Original Message-----
>> >> From: dispatch-bounces@ietf.org
>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> >> Sent: Monday, June 28, 2010 6:38 PM
>> >> To: DISPATCH
>> >> Subject: [dispatch] VIPR - proposed charter version 3
>> >>
>> >> Hi all,
>> >>
>> >> Below, please find version 3 of the VIPR proposed charter.
>> >> It's been updated to reflect ML feedback, in particular
>> VAP has been
>> >> added and clarifications have been made with regards to impacts
>> >> (i.e., none) on existing PSTN interfaces.
>> >>
>> >> Regards,
>> >> Mary
>> >> (DISPATCH WG co-chair)
>> >>
>> >> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>> >> VIPR Charter Proposal (Version 3)
>> >>
>> >> WG Name: =A0Verification Involving PSTN Reachability (VIPR)
>> >>
>> >> There are two globally deployed address spaces for communications
>> >> used by more than a billion people daily - phone numbers and DNS
>> >> rooted address such as web servers and email addresses. The
>> >> inter-domain signaling design of SIP is primarily designed
>> for email
>> >> style addresses yet a large percentage of SIP deployments
>> mostly use
>> >> phone numbers for identifying users. The goal of this
>> working group
>> >> is to enable inter-domain communications over the the
>> Internet, using
>> >> protocols such as SIP, while still allowing people to use phone
>> >> numbers to identify the person with whom they wish to communicate.
>> >>
>> >> The VIPR WG will address this problem by developing a peer to peer
>> >> based approach to finding domains that claim to be
>> responsible for a
>> >> given phone number and validation protocols to ensure a reasonable
>> >> likelihood that a given domain actually is responsible for
>> the phone
>> >> number. In this context, "responsible" means an administrative
>> >> domain, which is at least one of the domains, to which a
>> PSTN call to
>> >> this phone number would be routed. Once the domain responsible for
>> >> the phone number is found, existing protocols, such as SIP, can be
>> >> used for inter-domain communications.
>> >>
>> >> Some validation protocols may be based on knowledge
>> gathered around a
>> >> PSTN call; for example, the ability to prove a call was
>> received over
>> >> the PSTN based on start and stop times.
>> >> Other validation schemes, such as examining fingerprints or
>> >> watermarking of PSTN media to show that a domain received a
>> >> particular PSTN phone call, may also be considered by the working
>> >> group. This validation will be accomplished using publicly
>> available
>> >> open interfaces to the PSTN, so the validation can be performed by
>> >> any domain wishing to participate. =A0The WG will select and
>> >> standardize at least one validation scheme.
>> >>
>> >> The validation mechanism requires a domain to gather and maintain
>> >> information related to PSTN calls. =A0This information is
>> used by call
>> >> agents such as phones, SBCs and IP PBXs to route calls.
>> The WG will
>> >> define a client-server protocol between these call agents and the
>> >> entity within a domain that maintains the information.
>> >>
>> >> To help mitigate SPAM issues when using SIP between
>> domains, the WG
>> >> will define a mechanism to enable one domain to check that
>> incoming
>> >> SIP messages are coming from a validated phone number. =A0A phone
>> >> number is considered validated if it is coming from a
>> domain to which
>> >> the calling domain had previously successfully placed a
>> PSTN call.
>> >> The working group will define new SIP headers and option tags, as
>> >> necessary, to enable this.
>> >>
>> >> The essential characteristic of VIPR is establishing
>> authentication
>> >> by PSTN reachability when it is not possible to use a direct
>> >> reference to ENUM databases or other direct assertions of
>> PSTN number
>> >> ownership. Elements such as public ENUM easily coexist
>> with VIPR but
>> >> no direct interaction with ENUM will be required. =A0The
>> solution set
>> >> defined by this WG will be incrementally deployable using only
>> >> existing interfaces to the PSTN. =A0No changes will be required to
>> >> existing PSTN capabilities, no new database access is
>> needed nor is
>> >> any new support from PSTN service providers required.
>> >>
>> >> The problem statement and some possible starting points
>> for solutions
>> >> are further discussed in the following internet drafts which shall
>> >> form the bases of the WG documents:
>> >>
>> >> draft-rosenberg-dispatch-vipr-overview
>> >> draft-rosenberg-dispatch-vipr-reload-usage
>> >> draft-rosenberg-dispatch-vipr-pvp
>> >> draft-rosenberg-dispatch-vipr-sip-antispam
>> >> draft-rosenberg-dispatch-vipr-vap
>> >>
>> >> The working group will carefully coordinate with the
>> security area,
>> >> O&M area, as well as the appropriate RAI WGs such as sipcore and
>> >> p2psip.
>> >> _______________________________________________
>> >> dispatch mailing list
>> >> dispatch@ietf.org
>> >> https://www.ietf.org/mailman/listinfo/dispatch
>> >>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From dromasca@avaya.com  Wed Jun 30 08:33:25 2010
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Mary Barnes" <mary.ietf.barnes@gmail.com>
Cc: DISPATCH <dispatch@ietf.org>, IETF-Discussion list <ietf@ietf.org>
Subject: Re: [dispatch] VIPR - proposed charter version 3
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It looks to me that one can imagine 'centralized' solutions which are
also based on reusing SIP related functionality developed in RAI. I
would rather not close such an option and allow the WG a window of
opportunity in which alternate solutions that could meet the same goals
can be presented. =20

Dan


> -----Original Message-----
> From: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]=20
> Sent: Wednesday, June 30, 2010 6:24 PM
> To: Romascanu, Dan (Dan)
> Cc: DISPATCH; IETF-Discussion list
> Subject: Re: [dispatch] VIPR - proposed charter version 3
>=20
> Hi Dan,
>=20
> The term peer to peer is intended to exclude mechanisms that=20
> would use a central repository for the information:  This was=20
> discussed in an earlier thread:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg02027.html
>=20
> In one sense it is a solution, however, in another sense it=20
> is reusing SIP related functionality defined in RAI and thus=20
> is in a similar vein as specifying the use of SIP in a charter.
>=20
> Thanks,
> Mary.
>=20
> On Wed, Jun 30, 2010 at 5:42 AM, Romascanu, Dan (Dan)=20
> <dromasca@avaya.com> wrote:
> >> The VIPR WG will address this problem by developing a peer to peer=20
> >> based approach to finding domains that claim to be=20
> responsible for a=20
> >> given phone number and validation protocols to ensure a reasonable=20
> >> likelihood that a given domain actually is responsible for=20
> the phone=20
> >> number.
> >
> > Hi,
> >
> > Clarification question. What exactly means 'peer to peer=20
> based approach'
> > and what kind of approaches are excluded by having this in=20
> the charter.
> > Does 'approach' mean solution? If so why does a specific type of=20
> > solution need to be agreed in the charter, while all we=20
> have at hand=20
> > at this point are individual contribution I-Ds that describe the=20
> > 'problem statement and some possible starting points for solutions'?
> >
> > Thanks and Regards,
> >
> > Dan
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: Monday, June 28, 2010 8:38 PM
> >> To: DISPATCH
> >> Subject: [dispatch] VIPR - proposed charter version 3
> >>
> >
>=20

From mary.ietf.barnes@gmail.com  Wed Jun 30 08:47:12 2010
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
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Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Dan,

One of the starting points for this work is that more centralized
solutions to this problem have been defined (and implemented), yet
they have not been successful.  Per one of Christer's earllier
comments, we will add some text with reference to such and thus
substantiate why this group is focusing on a p2p based solution.

Thanks,
Mary.

On Wed, Jun 30, 2010 at 10:33 AM, Romascanu, Dan (Dan)
<dromasca@avaya.com> wrote:
> It looks to me that one can imagine 'centralized' solutions which are
> also based on reusing SIP related functionality developed in RAI. I
> would rather not close such an option and allow the WG a window of
> opportunity in which alternate solutions that could meet the same goals
> can be presented.
>
> Dan
>
>
>> -----Original Message-----
>> From: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
>> Sent: Wednesday, June 30, 2010 6:24 PM
>> To: Romascanu, Dan (Dan)
>> Cc: DISPATCH; IETF-Discussion list
>> Subject: Re: [dispatch] VIPR - proposed charter version 3
>>
>> Hi Dan,
>>
>> The term peer to peer is intended to exclude mechanisms that
>> would use a central repository for the information: =A0This was
>> discussed in an earlier thread:
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg02027.html
>>
>> In one sense it is a solution, however, in another sense it
>> is reusing SIP related functionality defined in RAI and thus
>> is in a similar vein as specifying the use of SIP in a charter.
>>
>> Thanks,
>> Mary.
>>
>> On Wed, Jun 30, 2010 at 5:42 AM, Romascanu, Dan (Dan)
>> <dromasca@avaya.com> wrote:
>> >> The VIPR WG will address this problem by developing a peer to peer
>> >> based approach to finding domains that claim to be
>> responsible for a
>> >> given phone number and validation protocols to ensure a reasonable
>> >> likelihood that a given domain actually is responsible for
>> the phone
>> >> number.
>> >
>> > Hi,
>> >
>> > Clarification question. What exactly means 'peer to peer
>> based approach'
>> > and what kind of approaches are excluded by having this in
>> the charter.
>> > Does 'approach' mean solution? If so why does a specific type of
>> > solution need to be agreed in the charter, while all we
>> have at hand
>> > at this point are individual contribution I-Ds that describe the
>> > 'problem statement and some possible starting points for solutions'?
>> >
>> > Thanks and Regards,
>> >
>> > Dan
>> >
>> >
>> >> -----Original Message-----
>> >> From: dispatch-bounces@ietf.org
>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> >> Sent: Monday, June 28, 2010 8:38 PM
>> >> To: DISPATCH
>> >> Subject: [dispatch] VIPR - proposed charter version 3
>> >>
>> >
>>
>

From ron.even.tlv@gmail.com  Wed Jun 30 09:04:07 2010
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Mary Barnes'" <mary.ietf.barnes@gmail.com>, "'Romascanu, Dan \(Dan\)'" <dromasca@avaya.com>
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Mary,
When I read the charter it is not clear why from the first paragraph you
deduct the second paragraph.
If the first paragraph will say
" The goal of this working group is to enable inter-domain communications
over the Internet, using protocols such as SIP, while still allowing people
to use the called person assigned circuit switch phone number for
identifying the person with whom they wish to communicate. " 
It may look clearer.

Still this is based on the assumption that using other ways to assign and
authenticate SIP phone numbers using central databases is not catching up
because the carriers object to implement it and not due to any other reason
like preferences of enterprises IT to use PSTN for calls because of better
quality of service and manageability.

Roni Even

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Mary Barnes
> Sent: Wednesday, June 30, 2010 6:24 PM
> To: Romascanu, Dan (Dan)
> Cc: DISPATCH; IETF-Discussion list
> Subject: Re: [dispatch] VIPR - proposed charter version 3
> 
> Hi Dan,
> 
> The term peer to peer is intended to exclude mechanisms that would use
> a central repository for the information:  This was discussed in an
> earlier thread:
> http://www.ietf.org/mail-archive/web/dispatch/current/msg02027.html
> 
> In one sense it is a solution, however, in another sense it is reusing
> SIP related functionality defined in RAI and thus is in a similar vein
> as specifying the use of SIP in a charter.
> 
> Thanks,
> Mary.
> 
> On Wed, Jun 30, 2010 at 5:42 AM, Romascanu, Dan (Dan)
> <dromasca@avaya.com> wrote:
> >> The VIPR WG will address this problem by developing a peer to
> >> peer based approach to finding domains that claim to be
> >> responsible for a given phone number and validation protocols
> >> to ensure a reasonable likelihood that a given domain
> >> actually is responsible for the phone number.
> >
> > Hi,
> >
> > Clarification question. What exactly means 'peer to peer based
> approach'
> > and what kind of approaches are excluded by having this in the
> charter.
> > Does 'approach' mean solution? If so why does a specific type of
> > solution need to be agreed in the charter, while all we have at hand
> at
> > this point are individual contribution I-Ds that describe the
> 'problem
> > statement and some possible starting points for solutions'?
> >
> > Thanks and Regards,
> >
> > Dan
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
> >> Sent: Monday, June 28, 2010 8:38 PM
> >> To: DISPATCH
> >> Subject: [dispatch] VIPR - proposed charter version 3
> >>
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From mary.ietf.barnes@gmail.com  Wed Jun 30 09:21:14 2010
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Date: Wed, 30 Jun 2010 11:21:22 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [dispatch] VIPR - proposed charter version 3
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Hi Roni,

Comments inline below and some snipping of the thread.

Thanks,
Mary.

On Wed, Jun 30, 2010 at 11:03 AM, Roni Even <ron.even.tlv@gmail.com> wrote:
> Mary,
> When I read the charter it is not clear why from the first paragraph you
> deduct the second paragraph.
> If the first paragraph will say
> " The goal of this working group is to enable inter-domain communications
> over the Internet, using protocols such as SIP, while still allowing peop=
le
> to use the called person assigned circuit switch phone number for
> identifying the person with whom they wish to communicate. "
> It may look clearer.
[MB] We could change "phone number" to "called person assigned circuit swit=
ch
phone number for identifying the person", but I think that "phone
number" is well
understood to be exactly what you want to describe it as.  [/MB]

>
> Still this is based on the assumption that using other ways to assign and
> authenticate SIP phone numbers using central databases is not catching up
> because the carriers object to implement it and not due to any other reas=
on
> like preferences of enterprises IT to use PSTN for calls because of bette=
r
> quality of service and manageability.
[MB] I'm not certain (personally) that the latter preferences you note
are as prevalent as you might suggest. I would still contend that the
larger issue is the former. However, if other folks share this
concern, we can certainly add that as well - it reflects equally as
being a non-technical reason for widespread implementation [/MB]
>
> Roni Even
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Mary Barnes
>> Sent: Wednesday, June 30, 2010 6:24 PM
>> To: Romascanu, Dan (Dan)
>> Cc: DISPATCH; IETF-Discussion list
>> Subject: Re: [dispatch] VIPR - proposed charter version 3
>>
>> Hi Dan,
>>
>> The term peer to peer is intended to exclude mechanisms that would use
>> a central repository for the information: =A0This was discussed in an
>> earlier thread:
>> http://www.ietf.org/mail-archive/web/dispatch/current/msg02027.html
>>
>> In one sense it is a solution, however, in another sense it is reusing
>> SIP related functionality defined in RAI and thus is in a similar vein
>> as specifying the use of SIP in a charter.
>>
>> Thanks,
>> Mary.
>>
>> On Wed, Jun 30, 2010 at 5:42 AM, Romascanu, Dan (Dan)
>> <dromasca@avaya.com> wrote:
>> >> The VIPR WG will address this problem by developing a peer to
>> >> peer based approach to finding domains that claim to be
>> >> responsible for a given phone number and validation protocols
>> >> to ensure a reasonable likelihood that a given domain
>> >> actually is responsible for the phone number.
>> >
>> > Hi,
>> >
>> > Clarification question. What exactly means 'peer to peer based
>> approach'
>> > and what kind of approaches are excluded by having this in the
>> charter.
>> > Does 'approach' mean solution? If so why does a specific type of
>> > solution need to be agreed in the charter, while all we have at hand
>> at
>> > this point are individual contribution I-Ds that describe the
>> 'problem
>> > statement and some possible starting points for solutions'?
>> >
>> > Thanks and Regards,
>> >
>> > Dan
>> >
>> >
>> >> -----Original Message-----
>> >> From: dispatch-bounces@ietf.org
>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Mary Barnes
>> >> Sent: Monday, June 28, 2010 8:38 PM
>> >> To: DISPATCH
>> >> Subject: [dispatch] VIPR - proposed charter version 3
>> >>
>> >
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>
>

From mary.ietf.barnes@gmail.com  Wed Jun 30 09:24:08 2010
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Subject: [dispatch] VIPR - proposed charter version 4
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Hi all,

Below, please find version 4 of the VIPR proposed charter. It's been
updated to reflect ML feedback over the past 24 hours.  The primary changes
are adding some background as to why other solutions are not sufficient and
thus a p2p approach is recommended.  Also, the deliverables are now described
textually rather than listing the current set of individual drafts
related to this problem.

Regards,
Mary
(DISPATCH WG co-chair)

============================================
VIPR Charter Proposal (Version 4)

WG Name:  Verification Involving PSTN Reachability (VIPR)

There are two globally deployed address spaces for communications used
by more than a billion people daily - phone numbers and DNS rooted
address such as web servers and email addresses. The inter-domain
signaling design of SIP is primarily designed for email style addresses
yet a large percentage of SIP deployments mostly use phone numbers for
identifying users, thus DNS lookups are not sufficient. Other approaches
such as public ENUM are not sufficient due to lack of widespread
deployment for non-technical reasons - i.e., the involvement of
government and service provider administrative entities needing to
approve and populate the registries.  Private federations have been
established to workaround the issue, however, that solution is not
scalable. The goal of this working group is to enable inter-domain
communications over the the Internet, using protocols such as SIP,
while still allowing people to use phone numbers to identify the person
with whom they wish to communicate.

The VIPR WG will develop a peer to peer based approach to finding
domains that claim to be responsible for a given phone number
to mitigate the involvement of centralized entities to avoid some of
the issues encountered by past attempts to support SIP inter-domain
communications. Validation protocols will be developed to ensure a
reasonable likelihood that a given domain actually is responsible for
the phone number. In this context, "responsible" means an
administrative domain, which is at least one of the domains, to which
a PSTN call to this phone number would be routed. Once the domain
responsible for the phone number is found, existing protocols, such
as SIP, can be used for inter-domain communications.

Some validation protocols may be based on knowledge gathered around a
PSTN call; for example, the ability to prove a call was received over
the PSTN based on start and stop times. Other validation schemes, such
as examining fingerprints or watermarking of PSTN media to show that a
domain received a particular PSTN phone call, may also be considered by
the working group. This validation will be accomplished using publicly
available open interfaces to the PSTN, so the validation can be
performed by any domain wishing to participate.  The WG will select and
standardize at least one validation scheme.

The validation mechanism requires a domain to gather and maintain
information related to PSTN calls.  This information is used by call
agents such as phones, SBCs and IP PBXs to route calls.  The WG will
define a client-server protocol between these call agents and the entity
within a domain that maintains the information.

To help mitigate SPAM issues when using SIP between domains, the WG will
define a mechanism to enable one domain to check that incoming SIP
messages are coming from a validated phone number.  A phone number is
considered validated if it is coming from a domain to which the calling
domain had previously successfully placed a PSTN call.  The working
group will define new SIP headers and option tags, as necessary, to
enable this.

The essential characteristic of VIPR is establishing authentication by
PSTN reachability when it is not possible to use a direct reference to
ENUM databases or other direct assertions of PSTN number
ownership. Elements such as public ENUM easily coexist with VIPR but no
direct interaction with ENUM will be required.  The solution set defined
by this WG will be incrementally deployable using only existing
interfaces to the PSTN.  No changes will be required to existing PSTN
capabilities, no new database access is needed nor is any new support
from PSTN service providers required.

The WG will produce the following deliverables:

1) A document describing the requirements, problem statement and
architectural approach to support SIP inter-domain communications.
2) A document describing the use of the p2psip protocol (RELOAD) as
applied to this problem space.
3) A document defining a scheme for validating the phone numbers using
publicly available open interfaces to the PSTN.
4) A document defining client-server protocol between call agents and the
entity within a domain that gathers and maintains information related to
PSTN calls.
5) A document describing a mechanism to mitigate SPAM issues.

The working group will carefully coordinate with the security area, O&M
area, as well as the appropriate RAI WGs such as sipcore and p2psip.

From James.Rafferty@dialogic.com  Wed Jun 30 10:56:06 2010
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From: James Rafferty <James.Rafferty@dialogic.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 30 Jun 2010 13:56:10 -0400
Thread-Topic: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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Cc: Elithy <Mohamed.Elithy@dialogic.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, Mohamed, "ietf@ietf.org" <ietf@ietf.org>
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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Hi,=20

My company has had the experience of deploying the pre-standard version of =
this PSTN to SIP UUI mechanism during the past 2 years. =20

As noted in the draft charter, UUI information is widely used on the PSTN f=
or applications such as offering input data into call centers and then pres=
erving that data as calls get transferred. =20

Since many contact centers are now built using SIP, but still have PSTN sub=
scribers (via SS7 ISUP or ISDN PRI), there is a need to be able to interwor=
k the user to user information from the PSTN side into SIP.  In our experie=
nce, the "Johnston uui draft" has accomplished this and we have several cus=
tomers that find this interworking to be valuable.  We also noted improveme=
nts from early drafts into the later ones in areas such as making better us=
e of the ITU-T protocol discriminator, thus enabling better interoperabilit=
y from the PSTN side into SIP.  =20

The major deficiency of the current draft is its non-standard status. Funct=
ionally, 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 suc=
h a mechanism. =20

James =20
-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Gonzalo Camarillo
Sent: Wednesday, June 30, 2010 2:51 AM
To: Paul Kyzivat
Cc: dispatch@ietf.org; DRAGE, Keith (Keith); ietf@ietf.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:
>=20
>=20
> DRAGE, Keith (Keith) wrote:
>> The UUI information is NOT ISUP. It is a transparent envelope to the ent=
ire ISDN, so it is not part of an ISDN protocol and therefore not part of a=
n 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 S=
IP-T are irrelevant in this case.
>>
>> The problem we do have though is that are legacy usages of this informat=
ion. In particular applications in PBXs carry it between themselves in ISDN=
 call establishment. The information itself is not standardised. If you wan=
t 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 existin=
g usage (which in my view would be a complete non-starter), but to allow th=
e transfer of the existing information when migrated to a SIP environment. =
The information transferred does not form part of SIP, and should not be st=
andardised as part of SIP.
>=20
> How many different mechanisms do we have to construct for the purpose of
> tunneling stuff over SIP?
>=20
> Its especially bad if the stuff is neither standardized nor negotiated.
> It then just provides more opportunity for non-interoperability.
>=20
> 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.
>=20
>         Thanks,
>         Paul
>=20
>> regards
>>
>> Keith
>>
>>
>>
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of
>>> bruno.chatras@orange-ftgroup.com
>>> Sent: Tuesday, June 29, 2010 1:00 PM
>>> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.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@ietf.org
>>>> [mailto:dispatch-bounces@ietf.org] De la part de Gonzalo Camarillo
>>>> Envoy=E9 : mardi 29 juin 2010 13:03 =C0 : 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@cisco.com>
>>>> To: iesg@ietf.org <iesg@ietf.org>
>>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>
>>> Robert Sparks
>>>>> <rjsparks@nostrum.com>
>>>>>
>>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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
>>>>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>>>>>
>>>>> Sep 10 - Problem statement and requirements document to IESG
>>>>> (Informational)
>>>>> Mar 11 - SIP call control UUI specification to IESG (PS)
>>>>> _______________________________________________
>>>>> IETF-Announce mailing list
>>>>> IETF-Announce@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>>
>>>> Cullen Jennings
>>>> For corporate legal information go to:
>>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

_______________________________________________
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From pkyzivat@cisco.com  Wed Jun 30 11:19:49 2010
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, Mohamed Elithy <Mohamed.Elithy@dialogic.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "ietf@ietf.org" <ietf@ietf.org>
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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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@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: Wednesday, June 30, 2010 2:51 AM
> To: Paul Kyzivat
> Cc: dispatch@ietf.org; DRAGE, Keith (Keith); ietf@ietf.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@ietf.org
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of
>>>> bruno.chatras@orange-ftgroup.com
>>>> Sent: Tuesday, June 29, 2010 1:00 PM
>>>> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.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@ietf.org
>>>>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>>>>> To: iesg@ietf.org <iesg@ietf.org>
>>>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>
>>>> Robert Sparks
>>>>>> <rjsparks@nostrum.com>
>>>>>>
>>>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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)
>>>>>> _______________________________________________
>>>>>> IETF-Announce mailing list
>>>>>> IETF-Announce@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>>> Cullen Jennings
>>>>> For corporate legal information go to:
>>>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From pkyzivat@cisco.com  Wed Jun 30 16:20:30 2010
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Date: Wed, 30 Jun 2010 19:20:33 -0400
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To: Mohamed Elithy <Mohamed.Elithy@dialogic.com>
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, "dispatch@ietf.org" <dispatch@ietf.org>, "ietf@ietf.org" <ietf@ietf.org>, James Rafferty <James.Rafferty@dialogic.com>
Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
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Mohamed Elithy wrote:
> Hi!
> We are currently adding all 3 parameters (encoding, content, and purpose) in case of an ISDN --> SIP initiated call with their specified default values defined in section 5 of the draft.
> 
> As far as the formatting, as the draft specifies the first byte of the header value is the protocol discriminator. we found that the best way to satisfy multiple customers dealing with different types of encoding is to pass the protocol discriminator along with the raw byte stream from the UUI IE on the ISDN side to the UUI header on the SIP side. This way the SIP AS or any other SIP entity receiving that header can decide how to decode the UUI info based on the value of the protocol discriminator.

I got all that. My question was about what specific value(s?) of the 
protocol discriminator are you using, and how do they correspond to the 
rest of the value.

The same question would apply if sip were not involved at all - if the 
call were an ISDN call.

Is there a broad based convention that most call center sw will use a 
particular encoding?

Or is it well known that certain commercial products use certain encodings?

Or is this just a private side arrangement configured between the caller 
and the callee in a specific deployment?

	Thanks,
	Paul

> Hope this helps.
> 
> Regards,
> -Mohamed
> 
> Mohamed Elithy | Engineering Manager/Architect | Dialogic Inc. | mohamed.elithy@dialogic.com | 508-862-3722 W | 508-364-5526 M
> 
> This e-mail is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. No waiver of privilege, confidence or otherwise is intended by virtue of communication via the internet. Any unauthorized use, dissemination or copying is strictly prohibited. If you have received this e-mail in error, or are not named as a recipient, please immediately notify the sender and destroy all copies of this e-mail.
> 
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, June 30, 2010 2:20 PM
> To: James Rafferty
> Cc: Gonzalo Camarillo; dispatch@ietf.org; DRAGE, Keith (Keith); ietf@ietf.org; Mohamed Elithy
> Subject: Re: [dispatch] Fwd: Re: WG Review: Call Control UUI for SIP (cuss)
> 
> 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@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>> Sent: Wednesday, June 30, 2010 2:51 AM
>> To: Paul Kyzivat
>> Cc: dispatch@ietf.org; DRAGE, Keith (Keith); ietf@ietf.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@ietf.org
>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of
>>>>> bruno.chatras@orange-ftgroup.com
>>>>> Sent: Tuesday, June 29, 2010 1:00 PM
>>>>> To: Gonzalo.Camarillo@ericsson.com; dispatch@ietf.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@ietf.org
>>>>>> [mailto:dispatch-bounces@ietf.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@cisco.com>
>>>>>> To: iesg@ietf.org <iesg@ietf.org>
>>>>>> CC: IETF Discussion Mailing List <ietf@ietf.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@ietf.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.Camarillo@ericsson.com>
>>>>> Robert Sparks
>>>>>>> <rjsparks@nostrum.com>
>>>>>>>
>>>>>>> Real-time Applications and Infrastructure Area Advisor:
>>>>>>>  Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.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)
>>>>>>> _______________________________________________
>>>>>>> IETF-Announce mailing list
>>>>>>> IETF-Announce@ietf.org
>>>>>>> https://www.ietf.org/mailman/listinfo/ietf-announce
>>>>>> Cullen Jennings
>>>>>> For corporate legal information go to:
>>>>>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> dispatch mailing list
>>>>>> dispatch@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>>
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> 
> 
