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This is a reminder, sent out once a month, about your ietf.org mailing
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From simple-bounces@ietf.org  Wed Sep  1 16:30:14 2004
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From: Robert Sparks <RjS@xten.com>
Subject: Re: [Simple] Minutes/proceedings - IETF60
Date: Wed, 1 Sep 2004 14:47:14 -0500
To: Robert Sparks <rsparks@dynamicsoft.com>
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Folks,

I received no comments on these draft minutes. I'm submitting them
as-is to the proceedings today.

Thanks,

RjS

On Aug 19, 2004, at 9:02 AM, Robert Sparks wrote:

> The slides from San Diego are on softarmor at
> http://www.softarmor.com/simple/meets/ietf60/slides/
> and in the draft proceedings at
> http://www.ietf.org/proceedings/04aug/index.html
>
> Draft minutes are attached.
>
> Please provide feedback/correction no later than 8/31.
>
> Thanks,
>
> RjS
>
>
> <minutes>_______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple



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From simple-bounces@ietf.org  Wed Sep  1 16:36:18 2004
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Subject: RE: [Simple] MSRP: Contributor ID
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From: "Orit Levin" <oritl@microsoft.com>
To: "Adam Roach" <adam@nostrum.com>, <hisham.khartabil@nokia.com>
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A clarification to the clarification.

We are not debating "a redundant mechanism for doing the same thing".
I don't think that there has been a doubt that MSRP entities MUST
support CPIM and MUST be able to receive and process it.

The ongoing debate is whether
- the CPIM technique can work for both centralized conferencing and
peer-to-peer security/privacy in the same time
OR
- MSRP needs to define additional means for conferencing support (in
order to be able to use CPIM for its original purpose!)

Orit.
=20
> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Tuesday, August 31, 2004 4:39 PM
> To: hisham.khartabil@nokia.com
> Cc: simple@ietf.org
> Subject: Re: [Simple] MSRP: Contributor ID
>=20
> hisham.khartabil@nokia.com wrote:
>=20
> >- Rohan will write up a separate internet draft describing=20
> the header approach with its security considerations.
> >
> >- Robert will kick off an internet draft describing the body=20
> approach (be it CPIM or something else) and its security=20
> considerations.
> >
>=20
> Just to be clear: Section 4.1.1 of RFC 2779 absolutely,=20
> unambiguously, and without recourse for appeal binds SIMPLE=20
> to support CPIM in MSRP.=20
> Section 4.1.2 similarly requires us to be able to identify=20
> the sender.=20
> These mechansims, simply put, must be in MRSP, or it dies in=20
> IESG review.
>=20
> Full stop.
>=20
> So, I want to clear up any misunderstanding that this might=20
> be an either-or decision -- an impression Hisham's note may=20
> have given some particpants.
>=20
> The support of CPIM _will_ be part of MSRP, it _will_ contain=20
> mechanisms for identifying the sender, and there's not a=20
> thing any member of this working group can do about it.
>=20
> At this point, we're just debating whether there will be a=20
> redundant mechanism for doing the same thing.
>=20
> /a
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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From simple-bounces@ietf.org  Thu Sep  2 09:14:18 2004
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        Internet Architecture Board <iab@iab.org>,
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Subject: [Simple] Protocol Action: 'Indication of Message Composition for 
 Instant Messaging' to Proposed Standard 
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The IESG has approved the following document:

- 'Indication of Message Composition for Instant Messaging '
   <draft-ietf-simple-iscomposing-03.txt> as a Proposed Standard

This document is the product of the SIP for Instant Messaging and Presence 
Leveraging Extensions Working Group. 

The IESG contact persons are Ted Hardie and Scott Hollenbeck.

Technical Summary

This document extends the SIMPLE IM system, defining a new status
message content type and XML namespace.  These will be used
to relay information during an IM conversation that indicates
that one party is composing a message and conveys information
about the information being composed  The status message can
indicate the composition of a message of any type, including text,
voice or video.  
 
Working Group Summary
 
The working group came to consensus on this document, and this feature
is widely implemented in existing IM systems, so its utility is 
well understood.
 
Protocol Quality
 
This document was reviewed for the IESG by Ted Hardie.


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Subject: [Simple] RFC 3856 on A Presence Event Package for the Session
	Initiation Protocol (SIP)
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3856

        Title:      A Presence Event Package for the Session
                    Initiation Protocol (SIP)
        Author(s):  J. Rosenberg
        Status:     Standards Track
        Date:       August 2004
        Mailbox:    jdrosen@dynamicsoft.com
        Pages:      27
        Characters: 62956
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-simple-presence-10.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3856.txt


This document describes the usage of the Session Initiation Protocol
(SIP) for subscriptions and notifications of presence.  Presence is
defined as the willingness and ability of a user to communicate with
other users on the network.  Historically, presence has been limited
to "on-line" and "off-line" indicators; the notion of presence here
is broader.  Subscriptions and notifications of presence are supported
by defining an event package within the general SIP event
notification framework.  This protocol is also compliant with the
Common Presence Profile (CPP) framework.

This document is a product of the SIP for Instant Messaging and
Presence Leveraging Extensions Working Group of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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A new Request for Comments is now available in online RFC libraries.


        RFC 3857

        Title:      A Watcher Information Event Template-Package for
                    the Session Initiation Protocol (SIP)
        Author(s):  J. Rosenberg
        Status:     Standards Track
        Date:       August 2004
        Mailbox:    jdrosen@dynamicsoft.com
        Pages:      20
        Characters: 46221
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-simple-winfo-package-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3857.txt


This document defines the watcher information template-package for the
Session Initiation Protocol (SIP) event framework.  Watcher
information refers to the set of users subscribed to a particular
resource within a particular event package.  Watcher information
changes dynamically as users subscribe, unsubscribe, are approved, or
are rejected.  A user can subscribe to this information, and therefore
learn about changes to it.  This event package is a template-package
because it can be applied to any event package, including itself.

This document is a product of the SIP for Instant Messaging and
Presence Leveraging Extensions Working Group of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
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A new Request for Comments is now available in online RFC libraries.


        RFC 3858

        Title:      An Extensible Markup Language (XML) Based Format
                    for Watcher Information
        Author(s):  J. Rosenberg
        Status:     Standards Track
        Date:       August 2004
        Mailbox:    jdrosen@dynamicsoft.com
        Pages:      13
        Characters: 26416
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-simple-winfo-format-04.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3858.txt


Watchers are defined as entities that request (i.e., subscribe to)
information about a resource.  There is fairly complex state
associated with these subscriptions.  The union of the state for all
subscriptions to a particular resource is called the watcher
information for that resource.  This state is dynamic, changing as
subscribers come and go.  As a result, it is possible, and indeed
useful, to subscribe to the watcher information for a particular
resource.  In order to enable this, a format is needed to describe the
state of watchers on a resource.  This specification describes an
Extensible Markup Language (XML) document format for such state.

This document is a product of the SIP for Instant Messaging and
Presence Leveraging Extensions Working Group of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
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From simple-bounces@ietf.org  Thu Sep  2 17:09:04 2004
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Subject: [Simple] Partial Publish as WG item
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OK - there appear to be several people willing to comment on this 
mechanism now and
rough consensus to take it on as a working group item, with
draft-lonnfors-simple-partial-publish-01 as a starting point.

Those folks who have chimed in with explicit support and concerns: we 
will be
  calling on you to review and comment.

Thanks,

RjS



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From simple-bounces@ietf.org  Tue Sep  7 08:08:16 2004
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From: "Faisal Adeem Siddiqui" <faysal@ajou.ac.kr>
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Subject: [Simple] Regarding the SIP register messages in SIP for Networked
	Appliances
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This is a multi-part message in MIME format.

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------=_NextPart_000_0343_01C4951E.07F9DEB0
Content-Type: text/plain;
	charset="ks_c_5601-1987"
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I want to know why the To and From addresses are same in this (see =
below) Register message ? Do we really need the "To" address ? Shouldn't =
the "To" address be "registrar@home.net" ?=20
Please explain.

REGISTER registrar@home.net
To: [slp:/d=3Dalarmclock, r=3Dbedroom,u=3Dstanm]@ua.stan.home.net
From: [slp:/d=3Dalarmclock, r=3Dbedroom,u=3Dstanm]@ua.stan.home.net=20
Content-type: application/ddp
[Device Address]

Thanks in advance.
Faisal.
------=_NextPart_000_0343_01C4951E.07F9DEB0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dks_c_5601-1987">
<META content=3D"MSHTML 6.00.2800.1458" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>I want to know why the To and =
From=20
addresses are same in this (see below)&nbsp;Register message ? Do we =
really=20
need&nbsp;the "To" address ?&nbsp;Shouldn't the "To" address be "<A=20
href=3D"mailto:registrar@home.net">registrar@home.net</A>" ? =
</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Please explain.</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>REGISTER <A=20
href=3D"mailto:registrar@home.net">registrar@home.net</A></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>To: [slp:/d=3Dalarmclock,=20
r=3Dbedroom,u=3Dstanm]@ua.stan.home.net</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>From: [slp:/d=3Dalarmclock,=20
r=3Dbedroom,u=3Dstanm]@ua.stan.home.net </FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Content-type:=20
application/ddp</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>[Device Address]</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Thanks in =
advance.</FONT></DIV><FONT=20
face=3D"Comic Sans MS" size=3D2>
<DIV>Faisal.</FONT></DIV></FONT></DIV></BODY></HTML>

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From simple-bounces@ietf.org  Tue Sep  7 08:51:05 2004
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Subject: RE: [Simple] Partial Publish as WG item
Date: Tue, 7 Sep 2004 15:41:31 +0300
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We will be producing a new version of the Patrial Publish internet draft =
as a WG item soon. Please review the current version and report any =
issues you find. The draft can be found at:

http://www.ietf.org/internet-drafts/draft-lonnfors-simple-partial-publish=
-01.txt

There is one open issue known to the authors, and that is the behaviour =
at the server when there are XML errors.

Regards,
Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Robert Sparks
> Sent: 02.September.2004 23:26
> To: simple@ietf.org
> Cc: Ted Hardie
> Subject: [Simple] Partial Publish as WG item
>=20
>=20
> OK - there appear to be several people willing to comment on this=20
> mechanism now and
> rough consensus to take it on as a working group item, with
> draft-lonnfors-simple-partial-publish-01 as a starting point.
>=20
> Those folks who have chimed in with explicit support and concerns: we=20
> will be
>   calling on you to review and comment.
>=20
> Thanks,
>=20
> RjS
>=20
>=20
>=20
> _______________________________________________
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To: Faisal Adeem Siddiqui <faysal@ajou.ac.kr>
Subject: Re: [Simple] Regarding the SIP register messages in SIP for Networked
	Appliances
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Faisal Adeem Siddiqui wrote:
> I want to know why the To and From addresses are same in this (see 
> below) Register message ? Do we really need the "To" address ? Shouldn't 
> the "To" address be "registrar@home.net <mailto:registrar@home.net>" ?
> Please explain.

- The To-URI identifies the address of record that is to be modified
  by the registration.

- The Request-URI identifies the registrar that is intended to process
  the registration. Often it is obtained by stripping the user part from
  the To-URI. But it is up the the UAC to do this, or otherwise
  determine what it is.

- The From-URI identifies the Address of record of the UAC that is
  sending the REGISTER message. It is certainly a very common case
  for this to be the same as the To-URI, but it need not be the
  case.

The technique of deriving the Request-URI is indeed odd and inconsistent
with what is typically done for other sip messages. I believe it is just
an artifact of the way that sip has evolved. But that is the way it is.

	Paul

> REGISTER registrar@home.net <mailto:registrar@home.net>
> To: [slp:/d=alarmclock, r=bedroom,u=stanm]@ua.stan.home.net
> From: [slp:/d=alarmclock, r=bedroom,u=stanm]@ua.stan.home.net
> Content-type: application/ddp
> [Device Address]
>  
> Thanks in advance.
> Faisal.
> 
> 
> ------------------------------------------------------------------------
> 
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From simple-bounces@ietf.org  Tue Sep  7 14:33:34 2004
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To: Orit Levin <oritl@microsoft.com>
Subject: Re: [Simple] MSRP: Contributor ID
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Sorry for delayed response - I've been on vacation, but am now back.

Orit Levin wrote:
> Paul,
> The answer to "why" in each of the cases relates to the fact that
> different cases belong to same architecture/possible deployment. Even
> when you see a solution to one case - you are unintentionally missing
> another unless you go through the exercise of defining the full
> conferencing architecture.
> 
> Instead, if we step back and try to look at the problem from the top,
> CPIM is a simple means for peer-to-peer private communications with
> non-trusted intermediates (e.g. GW, MCUs) potentially present in the
> middle. Using CPIM for communications between users and the
> intermediates themselves (and still being able to use CPIM for what it
> had been designed for) without defining standard ubiquitous means for
> distinguishing between the cases - is close to impossible.

I'm still not convinced of why the recipient needs to be able to 
distinguish the cases.

If a message is received, it could have:
- no indication of source. All that is known is the identification
   of the other end of the signaling path.

- it could have a single CPIM wrapper, unsigned, unencrypted,
   that identifies the source. The recipient doesn't know if this
   came from the actual sender, or from some intermediary. But
   does it matter? In either case it can't be authenticated.

- it could have a single CPIM wrapper, signed by the actual
   sender.

- it could have a single CPIM wrapper, signed by the mixer.

- it could have two CPIM wrappers, one signed by the sender,
   one signed by the mixer.

In all of the signed cases, what is the value in knowing via the 
protocol whether a signature came from the mixer or the sender? What is 
really important is who is signing and whether you can authenticate the 
signature. If you have multiple signatures that you trust, and the 
corresponding CPIM wrappers disagree about who the sender is, then you 
have to decide which to trust, or else just display them both. If you 
want to pick one, does it really help to know whether it came from the 
mixer or not?

> On the RTP note, RTP is different from MSRP in many aspects:

True. The analogy can only be stretched so far.

> - RTP doesn't use CPIM and as a result doesn't have this problem to
> start from :-)

True. That of course wasn't the analogy I was considering.

My point was that in a conference, the sender uses RTP/RTCP to talk to 
the mixer, including its capabilities for identifying the sender. Then 
the mixer uses RTP/RTCP to talk to the recipient, again using its 
capabilities for identifying the sender. There isn't one way for the 
sender to identify itself to the recipient and a different way for the 
mixer to represent its notion of the sender to the recipient.

An analogous same state of affairs could be achieved using CPIM, if the 
mixer refuses to accept encrypted messages that it cannot decrypt. It 
could then receive messages, remove any incoming CPIM wrapper, then add 
and sign a CPIM wrapper for the recipient.

> - SSRC is used as a stream identifier only. In the core RTP, there is no
> multiplexing inside the UDP connection (which is 1-to-1 with the RTP
> session). In MSRP, the from-path value is used as the part of the "MSRP
> session identifier" for multiplexing inside a TCP tunnel.

This is all true. But I don't see how it affects the discussion at hand.

> - in addition to the RTP "mixer" mode being implemented today, RTP also
> defines the "translator" mode - which is much closer to what happening
> in IM conferencing case. It this mode, the original SSRC is PRESERVED by
> translators (and the CSRC list is not created because it doesn't have
> any meaning at all).

This would be analogous to an MSRP mixer passing CPIM messages thru 
unmodified. Note that the description of translator mode in RFC 3550 
mentions that when it is used the recipient can't tell that a translator 
is present.

Note that RTP doesn't ever address the case where the sender doesn't 
provide CNAME info, or when it is wrong. Our case in MSRP of having the 
focus provide identification of the sender when the sender didn't 
provide that information is functionality not considered in RTP.

 > As it was stated earlier, voice and video
> conferencing vendors got away with it only because the user can guess
> the source by his/her voice or video. This doesn't work for IM unless an
> advanced conferencing protocol (e.g. XCON) is being used by ALL
> participants.

Agreed that it is more important to receive sender identification with 
MSRP than it is with RTP. But then we are free to work out what the 
extended needs are.

	Paul

> Orit.   
> 
> 
>>-----Original Message-----
>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>>Sent: Thursday, August 26, 2004 12:00 PM
>>To: Orit Levin
>>Cc: Ben Campbell; Adam Roach; Alan Johnston; simple@ietf.org
>>Subject: Re: [Simple] MSRP: Contributor ID
>>
>>
>>
>>Orit Levin wrote:
>>
>>> 
>>>
>>>>If
>>>>the message already had a cpim wrapper, then the focus 
>>>
>>could choose to 
>>
>>>>leave it alone, or put it inside yet another wrapper.
>>>>(You can have nested cpim wrappers.)
>>>>
>>>
>>>To accommodate the end-to-end encrypted CPIM, it will need to be 
>>>something like:
>>>If the message already had a cpim wrapper and the destination is 
>>>conference-aware, then the focus puts it inside yet another clear 
>>>wrapper (that is in order to provide mapping to conferencing 
>>>extensions).
>>>If the message already had a cpim wrapper and the destination is 
>>>conference-unaware (e.g. doesn't use XCON) leave it alone 
>>
>>(that is in 
>>
>>>order to not confuse the receiving client with conferencing 
>>>information vs. peer-to-peer information).
>>
>> >
>>
>>>This is VERY DIFFERENT from the existing today conferencing 
>>>architecture where the focus logic is NOT dependant on the 
>>>participant's ability to understand or actively support 
>>
>>conferencing extensions and protocols.
>>
>>I don't understand why you are splitting out different 
>>behaviors for conference aware and unaware participants.
>>
>>If the focus wants to ensure that messages are properly 
>>identified as to source, then it will either have to add a 
>>CPIM wrapper, or else inspect and approve one that is already 
>>present. If the message is encrypted e2e so that the focus 
>>can't decrypt, then it doesn't matter whether the encrypted 
>>body is CPIM or not. Either way the focus will have to wrap 
>>the encrypted body in a new CPIM wrapper.
>>
>>It can only avoid this if it doesn't feel it is necessary to 
>>identify the message sender. It will be up to the conference 
>>arch to decide about that. Maybe it will depend on whether 
>>the recipient is conference aware, maybe not.
>>
>>
>>>Yet "MY MAJOR PROBLEM" with using CPIM wrapper for 
>>
>>conferencing/focus 
>>
>>>can be summarised as following:
>>>There is no (well-defined simple) way in the CPIM header to 
>>>distinguish whether a particular CPIM is being used for 
>>
>>intermediate 
>>
>>>(e.g. focus,
>>>GW) vs. for peer-to-peer needs. A user may have very 
>>
>>different privacy 
>>
>>>policies regarding each.  In the case of conference-unaware 
>>
>>user - the 
>>
>>>UA will not even have the ability to distinguish between 
>>
>>the two cases.
>>
>>Why should there be? there isn't any way to do this for 
>>RTP/RTCP. There isn't any ability to have both kinds. And it 
>>is up to the mixer whether it preserves the IDs provided by 
>>the sender or puts in artificial ones. 
>>The recipient can't tell which is the case.
>>
>>
>>>Which identity will the user put in the requested CPIM header? The 
>>>private (for peer-to-peer) or the public for the focus use? How the 
>>>receiving participant (which can be conference unaware as 
>>
>>well) will 
>>
>>>undertsand what kind of identity he is looking at?
>>
>>If there are multiple unsigned and unencrypted identities, 
>>then the recipient should probably display them all, or maybe 
>>none of them if it is untrusting. If some or all of the 
>>identities are signed, then it could choose to only display 
>>ones signed by somebody it trusts.
>>
>>	Paul
>>
>>
> 
> 
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
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Subject: [Simple] Device priority
Date: Wed, 8 Sep 2004 16:01:47 +0200 
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Hello All,

I have a small question. 
Suppose I have two devices, a laptop and a mobile phone with the same
communication means ( they use same communication addresses). However, if
both devices are active (and registered), I would like to receive IM on my
laptop, and not on my phone. 

Is there a way to express to express the priority of the laptop over the
mobile phone for receiving IMs? I do not (immediately) find this in the
relevant RFC and IDs.

The way I understand the format of the presence document (RFC3863) is that
it groups a number of communication addresses (contact means + contact
address), and these communication addresses are decoupled from the devices
or user agents (which is not a bad idea I think).
 
Can someone enlighten my here?
Thanks in advance!

Greetings,

Stefan Goeman 


------_=_NextPart_001_01C495AC.615167D8
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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charset=3DUS-ASCII">
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5.5.2657.73">
<TITLE>[Simple] Device priority</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hello All,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I have a small question. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Suppose I have two devices, a laptop =
and a mobile phone with the same communication means ( they use same =
communication addresses). However, if both devices are active (and =
registered), I would like to receive IM on my laptop, and not on my =
phone. </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Is there a way to express to express =
the priority of the laptop over the mobile phone for receiving IMs? I =
do not (immediately) find this in the relevant RFC and IDs.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The way I understand the format of the =
presence document (RFC3863) is that it groups a number of communication =
addresses (contact means + contact address), and these communication =
addresses are decoupled from the devices or user agents (which is not a =
bad idea I think).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&nbsp;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Can someone enlighten my here?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Thanks in advance!</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Greetings,</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Stefan =
Goeman</FONT><BR>
<B></B><B></B>
</P>

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From simple-bounces@ietf.org  Wed Sep  8 11:41:43 2004
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From: Robert Sparks <RjS@xten.com>
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Cc: Ted Hardie <hardie@qualcomm.com>, hartmans@mit.edu
Subject: [Simple] Security Review of draft-ietf-simple-message-sessions-07
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Folks:

At Ted's request, Sam Hartman performed a security review of the MSRP 
base specification.
Sam's identified some things we need to do, and several things we need 
to talk about.
The chairs, our technical advisor, or the editors of these drafts will 
be sending messages to
the list responding to these points individually over the next few 
days, either with proposed text
or starting a discussion on the topic.

Please read this carefully and send your comments to the list.

Thanks,

RjS

Begin forwarded message:

>> This document reviews draft-ietf-simple-message-sessions-07.txt.  That
>> document specifies the MSRP protocol.
>>
>> 			  Protocol Overview
>>
>> The MSRP protocol is a deceptively simple protocol for transporting
>> messages in an instant messaging session.  It provides two operations:
>> a send request and a report request.  Each request can generate a
>> response.    Each endpoint is defined by a MSRP URL.  The URL contains
>> the typical  host and port, as well as a resource that identifies the
>> specific destination of a message.  For example  if Alice and Bob are
>> talking, they will both have MSRP URLs corresponding to their
>> identities.  MSRP specifies the optional use of TLS for transport
>> level protection.
>>
>> However the security picture is actually a lot more complicated.
>> First, the knowledge of the MSRP URL serves as authentication and
>> authorization information.   Since clients do not typically have TLS
>> certificates, a server knows it is talking to the right party based on
>> the URL that party sends to and possibly based on the response URL
>> listed in the message.    The URLs are used for authorization in that
>> layers above MSRP deal with preventing spam  and so it is important
>> that only parties authorized to send messages have the MSRP URL.   So
>> MSRP URLs need to be chosen with a strong random component and if
>> message integrity  is desired, they need to be
>> transported with confidentiality.
>> SIP, the layer above MSRP is responsible for mapping SIP or CPIM URLs
>> into MSRP URLs to set up a message session.  This mapping is security
>> critical because users are probably going to add SIP or CPIM URLs to
>> their contact lists.  So, to securely establish a connection, the
>> combination of SIP and MSRP must securely map the user-entered URL to
>> an MSRP URL.
>>
>> MSRP also involves S/MIME and CPIM identities.  If S/MIME is used to
>> get end-to-end encryption then the names of the chosen certificates
>> must be verified.  If message/cpim entities are sent, then the from
>> and to fields in these messages must be matched.  Typically the cpim
>> identities will be instant inbox URLs although perhaps they could also
>> be SIP URLs in some cases.
>>
>> One final identity is important for MSRP security.  The MSRP
>> specification requires that if TLS is used, the server certificate be
>> for the hostname in the destination MSRP URL.  This requirement for
>> the TLS identity has important security implications.
>>
>>
>> 			Security Requirements
>>
>> The draft does not specify a set of security requirements.  IT seems
>> reasonable to refer back to RFC 2779 for security requirements for
>> this protocol.
>>
>> I explicitly choose to distinguish confidentiality protection at the
>> transport layer from end-to-end confidentiality/integrity protection.
>> RFC 2779 doesn't quite explicitly make this distinction, but it does
>> require both integrity protection and support for digital signatures.
>> This distinction seems reasonable for several reasons.  First, other
>> IM protocols including XMPP and APEX support both levels of
>> protection.  We've found with email that end-to-end encryption and
>> transport level encryption are both useful; it seems likely that we
>> will see the same thing with IM.
>>
>> I also introduce the concept of parties sharing a deployed
>> authentication infrastructure.  Ideally, we'd have some authentication
>> infrastructure we could use to exchange key material with anyone on
>> the Internet.  In practice, we do not yet have this level of security
>> integration.  However we should try to provide security in
>> environments where we can do so.  Note that it is not acceptable to
>> force people to deploy a new authentication infrastructure for your
>> protocol if they already have an adequate infrastructure for some
>> other purpose.  If you can work within that infrastructure, then you
>> significantly decrease deployment costs of making your protocol
>> secure.
>>
>> 1)  In order to provide integrity protection, the protocol must
>> describe how identifiers  entered by the user are securely mapped  to
>> identifiers  used in the protocol.  The protocol must describe
>> necessary verifications of these mappings that recipients of messages
>> need to perform  in order to make sure they all refer to the same 
>> principal.
>>
>> 2) When two parties share an appropriate authentication
>> infrastructure,then the protocol will provide a way for them to get
>> hop-by-hop confidentiality and integrity.
>>
>> 3) When two parties share an appropriate authentication
>> infrastructure, then the protocol will provide a mechanism for
>> end-to-end confidentiality and integrity.
>>
>> 4) Even when no authentication infrastructure is shared, the protocol
>> will provide  protection against passive attacks.
>>
>> 			 Security Evaluation
>>
>> This document does not define, nor as far as I can tell, reference a
>> concrete procedure for mapping an instant inbox (im: URL) to a SIP
>> URL.  If clients can enter in im: URLs, then the related protocols
>> must specify a mechanism for  securely performing this mapping.
>>
>> The protocol specifies how SDP is used to map SIP URLs to MSRP URLs.
>> I do not have enough information about practical SIP deployments to
>> know whether this mapping is secure in practice.    IT seems possible
>> to secure this mapping if the SDP information is encrypted.
>>
>> This document does not specify how to  validate the certificates
>> encrypted or signed messages.    How do I know that a particular
>> certificate corresponds to a particular principal?  Should it include
>> an im: URL, a sip: URL, an msrp: URL or some combination?  Where
>> should these addresses be placed?
>> If the MSRP protocol is used to transport message/cpim content, the
>> recipient needs to confirm that the sender address in the message/cpim
>> corresponds to the expected sender.  This is especially important in
>> the case of signed messages.   The protocol needs to specify a
>> procedure for securely mapping an im: URL found in  the message/cpim
>> to  the appropriate MSRP URL  to validate the sender.
>>
>> The assumption that certificates used for TLS authentication for MSRP
>> should belong to the host seems flawed.  This might be a reasonable
>> assumption for the relay mode, but seems incorrect for the
>> peer-to-peer mode of the protocol.  IN a peer-to-peer configuration,
>> there may be multiple instances of MSRP on a given host in
>> significantly different security domains.  It seems necessary to
>> authenticate MSRP, SIP ,r IM peers rather than to authenticate hosts.
>>
>>
>> In practice, MSRP does not provide transport security in the
>> peer-to-peer mode.  The protocol allows for the use of TLS.  However
>> the spec points out that MSRP peers are not expected to have TLS
>> certificates or stable hostnames for these certificates to be bound
>> to.  The lack of deployable transport security is a critical
>> deficiency in the peer-to-peer mode of
>>  of MSRP.  Transport security has proven an important part of securing
>> XMPP.    Most other message transport protocols in the IETF, including
>> IMAP, SMTP, XMPP and APEX support transport security.
>> The protocol provides support for S/MIME for end-to-end security.
>> This is adequate to meet requirement 3 if the certificate naming
>> issues are addressed.
>>
>> The protocol does not provide a mechanism to protect against passive
>> attacks unless sufficient authentication infrastructure is available
>> to use TLS.
>>
>> 			   Recommendations
>>
>> 1) Decide on procedures for mapping im: URLs to sip: URLs.  These
>>    procedures are not part of MSRP, but MSRP needs a normative
>>    reference to them.  Note that RFC 3861 does not really answer this
>>    question.  That specification tells  you what machine to talk to in
>>    order to deliver an instant message.  It does not tell you what
>>    sip: URL to present to that system.
>> 2) Clearly state guidelines for dealing with the identities of message
>>    recipients and senders in S/MIME messages and message/cpim
>>    messages.  Look at section 6 of draft-ietf-xmpp-e2e for examples of
>>    the types of things such guidelines need to say.
>>
>> 3)  Re-examine the decision to use msrp: URLs.  It is my understanding
>>     from the change history that previous versions of the draft simply
>>     used sip: URLs.  That would be much easier to deal with from a
>>     security standpoint.  If the working group does decide to use
>>     msrp: URLs, then  the mapping needs to be carefully considered.
>>
>> 4)  Reconsider the decision to split relay functionality into its own
>>     draft.  I agree that for many configurations it is impossible to
>>     get good transport security without relay operation.  Rather than
>>     relaxing the requirement for transport security, I believe that
>>     the working group should come to the conclusion that relay mode is
>>     an integral part of the protocol.   Also, the current split does
>>     not work well because authentication to a relay is different than
>>     authentication to a peer.  In authenticating to a relay, you are
>>     authenticating to a specific service running on a host.  However
>>     when you authenticate to a peer, you are authenticating to a
>>     specific IM principal.  The current draft does not properly handle
>>     this distinction  and I don't think a split proposal can easily do
>>     so.  Note that I have not evaluated the security of the relay
>>     proposal, but I have considered what security is possible to
>>     achieve given solutions to the relay problem.
>>
>> 5)  Support SASL authentication and security layers for transport
>>     security in addition to TLS.  This should be done for
>>     compatibility with the other IETF messaging protocols.  In
>>     addition, some SASL modes  such as digest-md5 or the user-to-user
>>     Kerberos mechanism could work well for peer-to-peer
>>     authentication.  Providing implementers with this flexibility
>>     would be good.
>>
>> 6) Support TLS in an anonymous mode  for protection against passive
>>    attacks.  It's easy to do this  given that you already support TLS
>>    and will provide  useful benefits.
>>
>> 7) Consider handling of messages that appear to come from the same
>>    sender but are delivered over paths with different security
>>    properties.  For example, Alice is in a conversation with
>>    im:bob@example.com over an MSRP session using both TLS and
>>    end-to-end S/MIME.  How should Alice's client handle a message
>>    coming in over a different MSRP session using a different MSRP URL
>>    for the sender if that sender also maps back to im:bob@example.com?
>>    When is this acceptable from a security standpoint?
>>
>>
>> 		     Architectural Considerations
>>
>> While reviewing a document for security considerations, one often runs
>> into issues unrelated to security.  I discovered two such issues with
>> this document:
>>
>> 1) No mechanism is provided for discovery of future extensions.
>>     However, section 12 anticipates that future IETF action may define
>>     such extensions.  Significant experience has shown me that you
>>     want at least minimal discovery in your protocol.  You want to be
>>     able to add a negotiation mechanism later and discover its
>>     presence.  I recommend that MSRP clearly specify behavior if a
>>     client  uses an undefined method.  I recommend that this behavior
>>     be sufficient  to discover whether the peer implements some method
>>     in the future.
>>
>> 2) Section 6 of RFC 3862 lists several application considerations that
>>    applications using RFC 3862 need to address.  RFC 3860 does not
>>    address these considerations so MSRP must do so in order to
>>    completely specify the use of message/cpim.


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From simple-bounces@ietf.org  Wed Sep  8 12:45:14 2004
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From: Robert Sparks <RjS@xten.com>
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Subject: [Simple] Proposed applicability statement for MSRP
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Folks -

One of the things that Sam's review highlights is the need for a more 
explicit applicability statement in
the MSRP specs. Here's a proposal for the text for such a statement. 
(We'll add references in the
places that need them when we fold the result of this conversation into 
the spec).

RjS

---------------------------------------------------------
Applicability Statement:

MSRP is not designed for use as a standalone protocol. MSRP MUST only 
be used in the context of a rendezvous mechanism meeting the following 
requirements:

    - The rendezvous mechanism MUST provide both MSRP URIs associated 
with
      an MSRP session to each of the participating endpoints. The 
rendezvous
      mechanism MUST implement mechanisms to provide these URIs securely 
-
      they MUST NOT be made available to an untrusted third party or be 
easily discoverable.

  - The rendezvous mechanism MUST provide mechanisms for the negotiation 
of any
     supported MSRP extensions that are not backwards compatible.

    - The rendezvous mechanism MUST be able to natively transport im: 
URIs or automatically
      translate im: URIs into the addressing identifiers of the 
rendezvous protocol.

To use a rendezvous mechanism with MSRP, an RFC must be prepared 
describing
how it exchanges MSRP URIs and meets these requirements listed here. 
This document
provides such a description for the use of MSRP in the context of 
SIP/SDP.

The Session Initiation Protocol (SIP) meets these requirements for a
rendezvous mechanism. The MSRP URIs are exchanged using the
Session Description Protocol (SDP) in an offer/answer exchange using
SIP. The exchanged SDP can also be used to negotiate MSRP extensions.
This SDP can be secured using any of the mechanisms available in SIP, 
including
using sips to ensure transport security across intermediaries and 
S/MIME for
end-to-end protection of the SDP entity. SIP can carry arbitrary URIs 
(including im: URIs)
in the Request-URI, and procedures are available to map im: URIs to 
sip: or sips: URIs.
  It is expected that initial deployments of MSRP will use SIP as its 
rendezvous mechanism.


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Subject: Re: [Simple] Device priority
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inline.

Goeman Stefan wrote:

> Hello All,
> 
> I have a small question.
> Suppose I have two devices, a laptop and a mobile phone with the same 
> communication means ( they use same communication addresses). However, 
> if both devices are active (and registered), I would like to receive IM 
> on my laptop, and not on my phone.
> 
> Is there a way to express to express the priority of the laptop over the 
> mobile phone for receiving IMs? I do not (immediately) find this in the 
> relevant RFC and IDs.

 From a SIP perspective, the devices would register with different 
capabilities. See RFC 3840, which includes parameters to describe 
exactly this kidn of thing (support for IM vs. not).

A presence server could then use this registration information to 
generate a presence document. In this case, it can be modelled a few 
ways (see the presence data model document, 
http://www.ietf.org/internet-drafts/draft-rosenberg-simple-presence-data-model-00.txt. 
The most natural in this case is to model this as two sepearate 
services, each running on a separate device.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From simple-bounces@ietf.org  Wed Sep  8 15:03:31 2004
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Sorry again for the week long lag in my responses - finally found some 
time today to unbury myself from email backlog.

Responses inline.

Dave Cridland wrote:

> On Fri Aug 27 13:46:51 2004, Jonathan Rosenberg wrote:
> 
>> Dave Cridland wrote:
>>
>>> I'm a little confused by XCAP's purpose - is it aimed at providing a
>>>  configuration protocol specifically for the needs of SIMPLE's 
>>> protocol suite, or is it aimed at providing generalized configuration
>>>  access?
> 
> 
> 
>>> 1) How do vendor-specific extensions work? (Say, if an addressbook or
>>>  bookmark collection were stored in XCAP, how would an application 
>>> add its own data to it?) Is this handled purely by namespaces, or 
>>> does that interfere with the Schemas defined?
>>
>>
>> Each of the schemas in an application usage defines points where
>> extensions can add additional information. That additional information
>> would come from a different namespace.
>>
>>
> Predefined points? How does the designer know where extension might be 
> required?

Thats part of the task of designing the schema. There are many examples 
throughout XML document designs.


>>
>> The following spec defines a format for differences in documents:
>> http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-package-02.txt
>>
>> These documents would get delivered in notifications, using a SIP event
>> package:
>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-04.txt 
>>
>>
>>
> Ah... So XCAP requires SIP to be of any real use?

As Joel pointed out, this is a matter of perspective. There are many 
envisioned uses where the change notifications are not needed.


>>> I'm thinking particularly of, for instance, storing bookmarks in
>>> XCAP, which would require some method to track which bookmarks had
>>> been visited recently - in other words, every bookmark usage would
>>> then invalidate the entire document. That's truly terrifying.
>>
>>
>> I don't follow what you intend to store in the server. The bookmark 
>> URL and some kind of indication of when it was last visited?
> 
> 
> Yes, exactly that. (And more, of course.)
> 
> It's a useful feature which allows the notion of bookmarks and history 
> to merge, somewhat.

I think that is achievable with XCAP.

> 
> 
> Having read through your comments a little more, I think I understand 
> where XCAP fits in a little better.
> 
> As I understand things, a typical XCAP client would:
> 
> 1) Bootstrap its initial configuration with a GET request for the area 
> that interested it from XCAP. (I'm unclear how it might find the XCAP 
> server, but finding your configuration server is always fun.)

THis is described for SIP elements in the above-mentioned document.

> 2) Possibly using this information, acquire SIP.

I don't know what it means to "acquire SIP". The client may or may not 
support SIP depending on what it needs to do.

> 3) Write any changes using XCAP PUT and DELETE. Since PUT invalidates 
> the cache, ignore any result other than success or failure for PUT.

In your specific example, I would expect the PUT to be made against the 
specific bookmark whose status is being updated, each of which would be 
a different URI. SO, a simple-minded XML doc might look like this:

<bookmarks>
   <site name="yahoo">
     <url>http://www.yahoo.com</url>
     <last-visited>Sep 8, 2004</last-visited>
   </site>
   <site name="ietf">
     <url>http://www.ietf.org</url>
     <last-visited>Sep 8, 2004</last-visited>
   </site>
</bookmarks>

If I visit the ietf site, the browser would do something like this:

PUT 
http://xcap.example.com/bookmarks/users/joe/index/~~/bookmarks/site[@name="ietf"]/last-visited

<last-visited>Sep 9, 2004</last-visited>




> 4) Listen constantly for updates via SIP.

Are you expecting multiple clients to be updating the same bookmark file?

> 5) On reconnect, use acquire SIP using cached data, and then use SIP to 
> resynchronize.

If you don't want SIP, the other chocie is that the client can cache the 
etag for the document, and on re-connect, use HEAD to check if it has 
changed. If not, do nothing. If so, pull the whole new bookmark file.


> 
> Is it not possible to handle the writes via SIP, which appears to be 
> required anyway? And if so, couldn't nearly all the complexity of XCAP 
> be dropped out to provide a very simple configuration bootstrap protocol?

SIP doesnt do anything near what XCAP does. As such, there would be no 
savings in complexity. Its also well outside of the scope of SIP.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Adding to what Jonathan said:

Goeman Stefan wrote:
> Hello All,
> 
> I have a small question.
> Suppose I have two devices, a laptop and a mobile phone with the same 
> communication means ( they use same communication addresses). However, 
> if both devices are active (and registered), I would like to receive IM 
> on my laptop, and not on my phone.

You say they use the same communication address. Since they are separate 
devices, they must have separate contact addresses. So it must be the 
Address of Record that is the same, which presumably is also the 
presentity address.

There is some other detail you didn't provide:

- Is the laptop limited to IM, or does it also do voice?
   (I presume the mobile phone does both.)

- Do they use the same contact address for IM and voice?
   (sip for both.) Or do they use separate addresses
   for voice and IM? (e.g. sip: for voice, im: for IM.)

> Is there a way to express to express the priority of the laptop over the 
> mobile phone for receiving IMs? I do not (immediately) find this in the 
> relevant RFC and IDs.
> 
> The way I understand the format of the presence document (RFC3863) is 
> that it groups a number of communication addresses (contact means + 
> contact address), and these communication addresses are decoupled from 
> the devices or user agents (which is not a bad idea I think).
> 
>  
> Can someone enlighten my here?
> Thanks in advance!

If the the laptop only does IM, then its contact can be assigned a 
higher priority than that the mobile phone contact(s). As Jonathan 
mentioned, capabilities can be used to describe that this one does IM 
and not voice.

If they both do IM & voice, and they each use a single sip contact it 
gets a bit more tricky. With just two tuples you can't describe that one 
is better for IM, but not better for voice. To achieve this effect, you 
would have to publish two tuples for the laptop - one with the contact 
at a high priority, with capabilities of IM and not voice. Another 
tuple, with the same contact address but a lower priority, could show 
capabilties for both IM and voice. The mobile phone could be described 
by either one or two tuples, with capabilties and priorities assigned to 
achieve the kind of prioritization you want.

If IM and voice are handled using different contact addresses then you 
are forced to have separate tuples anyway, and to describe the 
capabilities separately. And of course you can set the priorities as you 
wish.

In some cases a useful presence document can be derived from 
registration information (particularly if you use callee capabilties in 
the registration). But in registration you aren't permitted to have two 
entries with the same contact address. So you can't derive a presence 
document that has two tuples with the same contact from registration. 
Either you must separately publish the presence doc, or else you must 
hack the registration - using multiple addresses for the same device in 
order to register it twice. But this isn't a good idea because it can 
result in the same device getting tried twice for the same call.

	Paul



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Sam,

Thanks first of all for the good review of the protocol - you make a lot of
important points below, and I think we will definitely need to make some
changes to the MSRP specification as a result of your suggestions. Here's
some notes for starters. ;)

The most important of change, I think, is the applicability statement that
Robert sent out separately today. I think the root cause of a number of the
deficiencies you note below is that the reliance of MSRP on a higher-layer
rendezvous mechanism isn't as obvious as the authors hoped.

The mapping of MSRP URLs to SIP URIs is perhaps the foremost of the related
issues you raise (your recommendation #3). To really understand the reliance
of MSRP on SIP, you have to look at the way SIP and RTP interact for
establishing multimedia calls. In a VoIP scenario, for example, SDP contains
the IP and port to which RTP will be sent - this identifies an RTP endpoint.
An MSRP URL serves the same function as an RTP IP/port; it is not intended
to be a user identity as such, user identities are exchanged and authorized
and so on at the SIP layer. Talk of difficulties "mapping" SIP URIs to MSRP
URLs, therefore, I think, mistakes the applicability of MSRP a bit, just as
I'd say if someone suggested we needed a way to "map" RTP IP/ports to SIP
URIs. The relationship of a SIP URI to an MSRP URL is totally transient and
contingent, just as the relationship between a SIP URI and any given RTP
endpoint IP/port is contingent (I may use this phone today, but some
different phone tomorrow). What binds a SIP URI to an RTP IP/port,
temporarily, is SIP itself - a SIP INVITE is sent containing SDP that
advertises where to send RTP. That RTP endpoint is meaningful only for the
duration of the session resulting from the SDP offer/answer. The resulting
binding is just as secure as SIP is - if SIP is used with integrity
protection and confidentiality over the SDP body, then the binding between
the identity in SIP and the RTP endpoint is a secure binding. All of this
applies as well to MSRP. Without some SIP-like means of securely exchanging
endpoint identifiers, then yes, the relationship between an end user's
identity and an MSRP URL would be very uncertain. With the applicability
more firmly in mind, though, I don't think there is really any "mapping"
problem here.

Similarly, I think the rendezvous protocol can (and in the case of SIP,
does) provide the sort of functionality that you'd get from something like
SASL (your recommendation #5). Digest auth is built into SIP, there are ways
of doing EAP (and all that EAP potentially brings with it) in SIP. More
important, establishing the identity of the participants in a session is
SIP's responsibility, not the responsibility of something at the RTP/MSRP
layer, given the binding established by SIP described above. While there are
many threats that motivate channel security at the RTP/MSRP layer, I think
TLS and S/MIME, collectively, can address those threats for MSRP, and they
are documented in the MSRP specification. SIP/SDP can serve as a vehicle to
deliver keying information (even self-signed certificates) between
endpoints, as it does in the sdescriptions method of exchanging keys for
sRTP (which has your bearing on recommendation #6). The statement that SASL
should be used for 'compatibility with the other IETF messaging protocols'
strikes me as a little odd, given that CPIM MSGFMT was designed to provide
precisely that, and was specified in the IMPP WG to allow compatibility
between IETF instant messaging systems based on different technologies. It's
also worth noting that interworking between various IETF messaging protocols
will almost certainly entail the presence of gateways, which suggests that
an end-to-end security solution like S/MIME MSGFMT would be more suitable
than a transport security system.

Does that make sense, and persuade? This isn't to say that I imagine
everything above to be obvious from the text of the draft today; we may need
to make further clarifications beyond the applicability statement to get
this across. But does the underlying story work, in your opinion?

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Robert Sparks [mailto:RjS@xten.com]
> Sent: Wednesday, September 08, 2004 8:21 AM
> To: simple@ietf.org
> Cc: Ted Hardie; hartmans@mit.edu
> Subject: [Simple] Security Review of
> draft-ietf-simple-message-sessions-07
> 
> 
> Folks:
> 
> At Ted's request, Sam Hartman performed a security review of the MSRP 
> base specification.
> Sam's identified some things we need to do, and several 
> things we need 
> to talk about.
> The chairs, our technical advisor, or the editors of these 
> drafts will 
> be sending messages to
> the list responding to these points individually over the next few 
> days, either with proposed text
> or starting a discussion on the topic.
> 
> Please read this carefully and send your comments to the list.
> 
> Thanks,
> 
> RjS
> 
> Begin forwarded message:
> 
> >> This document reviews 
> draft-ietf-simple-message-sessions-07.txt.  That
> >> document specifies the MSRP protocol.
> >>
> >> 			  Protocol Overview
> >>
> >> The MSRP protocol is a deceptively simple protocol for transporting
> >> messages in an instant messaging session.  It provides two 
> operations:
> >> a send request and a report request.  Each request can generate a
> >> response.    Each endpoint is defined by a MSRP URL.  The 
> URL contains
> >> the typical  host and port, as well as a resource that 
> identifies the
> >> specific destination of a message.  For example  if Alice 
> and Bob are
> >> talking, they will both have MSRP URLs corresponding to their
> >> identities.  MSRP specifies the optional use of TLS for transport
> >> level protection.
> >>
> >> However the security picture is actually a lot more complicated.
> >> First, the knowledge of the MSRP URL serves as authentication and
> >> authorization information.   Since clients do not 
> typically have TLS
> >> certificates, a server knows it is talking to the right 
> party based on
> >> the URL that party sends to and possibly based on the response URL
> >> listed in the message.    The URLs are used for 
> authorization in that
> >> layers above MSRP deal with preventing spam  and so it is important
> >> that only parties authorized to send messages have the 
> MSRP URL.   So
> >> MSRP URLs need to be chosen with a strong random component and if
> >> message integrity  is desired, they need to be
> >> transported with confidentiality.
> >> SIP, the layer above MSRP is responsible for mapping SIP 
> or CPIM URLs
> >> into MSRP URLs to set up a message session.  This mapping 
> is security
> >> critical because users are probably going to add SIP or 
> CPIM URLs to
> >> their contact lists.  So, to securely establish a connection, the
> >> combination of SIP and MSRP must securely map the 
> user-entered URL to
> >> an MSRP URL.
> >>
> >> MSRP also involves S/MIME and CPIM identities.  If S/MIME 
> is used to
> >> get end-to-end encryption then the names of the chosen certificates
> >> must be verified.  If message/cpim entities are sent, then the from
> >> and to fields in these messages must be matched.  
> Typically the cpim
> >> identities will be instant inbox URLs although perhaps 
> they could also
> >> be SIP URLs in some cases.
> >>
> >> One final identity is important for MSRP security.  The MSRP
> >> specification requires that if TLS is used, the server 
> certificate be
> >> for the hostname in the destination MSRP URL.  This requirement for
> >> the TLS identity has important security implications.
> >>
> >>
> >> 			Security Requirements
> >>
> >> The draft does not specify a set of security requirements. 
>  IT seems
> >> reasonable to refer back to RFC 2779 for security requirements for
> >> this protocol.
> >>
> >> I explicitly choose to distinguish confidentiality 
> protection at the
> >> transport layer from end-to-end confidentiality/integrity 
> protection.
> >> RFC 2779 doesn't quite explicitly make this distinction, 
> but it does
> >> require both integrity protection and support for digital 
> signatures.
> >> This distinction seems reasonable for several reasons.  
> First, other
> >> IM protocols including XMPP and APEX support both levels of
> >> protection.  We've found with email that end-to-end encryption and
> >> transport level encryption are both useful; it seems likely that we
> >> will see the same thing with IM.
> >>
> >> I also introduce the concept of parties sharing a deployed
> >> authentication infrastructure.  Ideally, we'd have some 
> authentication
> >> infrastructure we could use to exchange key material with anyone on
> >> the Internet.  In practice, we do not yet have this level 
> of security
> >> integration.  However we should try to provide security in
> >> environments where we can do so.  Note that it is not acceptable to
> >> force people to deploy a new authentication infrastructure for your
> >> protocol if they already have an adequate infrastructure for some
> >> other purpose.  If you can work within that 
> infrastructure, then you
> >> significantly decrease deployment costs of making your protocol
> >> secure.
> >>
> >> 1)  In order to provide integrity protection, the protocol must
> >> describe how identifiers  entered by the user are securely 
> mapped  to
> >> identifiers  used in the protocol.  The protocol must describe
> >> necessary verifications of these mappings that recipients 
> of messages
> >> need to perform  in order to make sure they all refer to the same 
> >> principal.
> >>
> >> 2) When two parties share an appropriate authentication
> >> infrastructure,then the protocol will provide a way for them to get
> >> hop-by-hop confidentiality and integrity.
> >>
> >> 3) When two parties share an appropriate authentication
> >> infrastructure, then the protocol will provide a mechanism for
> >> end-to-end confidentiality and integrity.
> >>
> >> 4) Even when no authentication infrastructure is shared, 
> the protocol
> >> will provide  protection against passive attacks.
> >>
> >> 			 Security Evaluation
> >>
> >> This document does not define, nor as far as I can tell, 
> reference a
> >> concrete procedure for mapping an instant inbox (im: URL) to a SIP
> >> URL.  If clients can enter in im: URLs, then the related protocols
> >> must specify a mechanism for  securely performing this mapping.
> >>
> >> The protocol specifies how SDP is used to map SIP URLs to 
> MSRP URLs.
> >> I do not have enough information about practical SIP deployments to
> >> know whether this mapping is secure in practice.    IT 
> seems possible
> >> to secure this mapping if the SDP information is encrypted.
> >>
> >> This document does not specify how to  validate the certificates
> >> encrypted or signed messages.    How do I know that a particular
> >> certificate corresponds to a particular principal?  Should 
> it include
> >> an im: URL, a sip: URL, an msrp: URL or some combination?  Where
> >> should these addresses be placed?
> >> If the MSRP protocol is used to transport message/cpim content, the
> >> recipient needs to confirm that the sender address in the 
> message/cpim
> >> corresponds to the expected sender.  This is especially 
> important in
> >> the case of signed messages.   The protocol needs to specify a
> >> procedure for securely mapping an im: URL found in  the 
> message/cpim
> >> to  the appropriate MSRP URL  to validate the sender.
> >>
> >> The assumption that certificates used for TLS 
> authentication for MSRP
> >> should belong to the host seems flawed.  This might be a reasonable
> >> assumption for the relay mode, but seems incorrect for the
> >> peer-to-peer mode of the protocol.  IN a peer-to-peer 
> configuration,
> >> there may be multiple instances of MSRP on a given host in
> >> significantly different security domains.  It seems necessary to
> >> authenticate MSRP, SIP ,r IM peers rather than to 
> authenticate hosts.
> >>
> >>
> >> In practice, MSRP does not provide transport security in the
> >> peer-to-peer mode.  The protocol allows for the use of 
> TLS.  However
> >> the spec points out that MSRP peers are not expected to have TLS
> >> certificates or stable hostnames for these certificates to be bound
> >> to.  The lack of deployable transport security is a critical
> >> deficiency in the peer-to-peer mode of
> >>  of MSRP.  Transport security has proven an important part 
> of securing
> >> XMPP.    Most other message transport protocols in the 
> IETF, including
> >> IMAP, SMTP, XMPP and APEX support transport security.
> >> The protocol provides support for S/MIME for end-to-end security.
> >> This is adequate to meet requirement 3 if the certificate naming
> >> issues are addressed.
> >>
> >> The protocol does not provide a mechanism to protect 
> against passive
> >> attacks unless sufficient authentication infrastructure is 
> available
> >> to use TLS.
> >>
> >> 			   Recommendations
> >>
> >> 1) Decide on procedures for mapping im: URLs to sip: URLs.  These
> >>    procedures are not part of MSRP, but MSRP needs a normative
> >>    reference to them.  Note that RFC 3861 does not really 
> answer this
> >>    question.  That specification tells  you what machine 
> to talk to in
> >>    order to deliver an instant message.  It does not tell you what
> >>    sip: URL to present to that system.
> >> 2) Clearly state guidelines for dealing with the 
> identities of message
> >>    recipients and senders in S/MIME messages and message/cpim
> >>    messages.  Look at section 6 of draft-ietf-xmpp-e2e for 
> examples of
> >>    the types of things such guidelines need to say.
> >>
> >> 3)  Re-examine the decision to use msrp: URLs.  It is my 
> understanding
> >>     from the change history that previous versions of the 
> draft simply
> >>     used sip: URLs.  That would be much easier to deal with from a
> >>     security standpoint.  If the working group does decide to use
> >>     msrp: URLs, then  the mapping needs to be carefully considered.
> >>
> >> 4)  Reconsider the decision to split relay functionality 
> into its own
> >>     draft.  I agree that for many configurations it is 
> impossible to
> >>     get good transport security without relay operation.  
> Rather than
> >>     relaxing the requirement for transport security, I believe that
> >>     the working group should come to the conclusion that 
> relay mode is
> >>     an integral part of the protocol.   Also, the current 
> split does
> >>     not work well because authentication to a relay is 
> different than
> >>     authentication to a peer.  In authenticating to a 
> relay, you are
> >>     authenticating to a specific service running on a 
> host.  However
> >>     when you authenticate to a peer, you are authenticating to a
> >>     specific IM principal.  The current draft does not 
> properly handle
> >>     this distinction  and I don't think a split proposal 
> can easily do
> >>     so.  Note that I have not evaluated the security of the relay
> >>     proposal, but I have considered what security is possible to
> >>     achieve given solutions to the relay problem.
> >>
> >> 5)  Support SASL authentication and security layers for transport
> >>     security in addition to TLS.  This should be done for
> >>     compatibility with the other IETF messaging protocols.  In
> >>     addition, some SASL modes  such as digest-md5 or the 
> user-to-user
> >>     Kerberos mechanism could work well for peer-to-peer
> >>     authentication.  Providing implementers with this flexibility
> >>     would be good.
> >>
> >> 6) Support TLS in an anonymous mode  for protection against passive
> >>    attacks.  It's easy to do this  given that you already 
> support TLS
> >>    and will provide  useful benefits.
> >>
> >> 7) Consider handling of messages that appear to come from the same
> >>    sender but are delivered over paths with different security
> >>    properties.  For example, Alice is in a conversation with
> >>    im:bob@example.com over an MSRP session using both TLS and
> >>    end-to-end S/MIME.  How should Alice's client handle a message
> >>    coming in over a different MSRP session using a 
> different MSRP URL
> >>    for the sender if that sender also maps back to 
> im:bob@example.com?
> >>    When is this acceptable from a security standpoint?
> >>
> >>
> >> 		     Architectural Considerations
> >>
> >> While reviewing a document for security considerations, 
> one often runs
> >> into issues unrelated to security.  I discovered two such 
> issues with
> >> this document:
> >>
> >> 1) No mechanism is provided for discovery of future extensions.
> >>     However, section 12 anticipates that future IETF 
> action may define
> >>     such extensions.  Significant experience has shown me that you
> >>     want at least minimal discovery in your protocol.  You 
> want to be
> >>     able to add a negotiation mechanism later and discover its
> >>     presence.  I recommend that MSRP clearly specify behavior if a
> >>     client  uses an undefined method.  I recommend that 
> this behavior
> >>     be sufficient  to discover whether the peer implements 
> some method
> >>     in the future.
> >>
> >> 2) Section 6 of RFC 3862 lists several application 
> considerations that
> >>    applications using RFC 3862 need to address.  RFC 3860 does not
> >>    address these considerations so MSRP must do so in order to
> >>    completely specify the use of message/cpim.
> 
> 
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From simple-bounces@ietf.org  Wed Sep  8 21:17:26 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Robert Sparks'" <RjS@xten.com>, simple@ietf.org
Subject: on relays (was RE: [Simple] Security Review of draft-ietf-simple-
	message-sessions-07)
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Sam,

Next topic. Relays (your recommendation #4).

While I recognize that other messaging protocols developed here in the IETF
have made relays an integral part of their operation, I'm very reluctant to
make relays mandatory for MSRP, as compelling as the transport security
argument may be.

First of all, let me be clear that I'm talking about MSRP relays, not
SIP-layer relays like SIP proxy servers. SIP proxies can and should play a
number of important roles in allowing MSRP endpoints to discover one
another, exchange SDP necessarily to the establishment of MSRP, provide
security properties (like hop-by-hop TLS, as described in RFC3261 26.3.2.2),
and implement various sorts of operator policies that are irrespective of
the -contents- of the MSRP session. All of that stuff is very important, has
to be provided by an intermediary, and is totally unrelated to whether or
not MSRP relays even exist.

The presence of relays in the operation of a protocol like RTP or MSRP is a
cause of latency, additional single points of failure, additional state
being kept in the network, and additional security concerns (because even
relays could misbehave) - all of the general problems that arise when you
don't follow the end-to-end model. In my view, the main reason why MSRP
relays exist at all is NATs - they facilitate NAT traversal. There are also
some other corner-cases, like IM logging in accordance with enterprise
policies, aggregating transport-layer connections between service providers,
and so on. But primarily, MSRP relays exist to compensate for a lack of
network-layer connectivity/addressibility between MSRP endpoints.

While it might seem attractive to say that MSRP endpoints could always use
TLS to connect to a relay that they trust, the same way we do this in SIP,
and thus guarantee that TLS can always be used for MSRP, there are many
levels of trust:

- You can trust a relay to forward your messages
- You can trust a relay not to modify your messages
- You can trust a relay  not to read your messages

Any relay that could pass the simplest empirical test has to forward your
messages, at least. But if you're concerned about a relay modifying or
reading your messages, you'll need to use something like S/MIME to prevent
that. S/MIME credentials (even self-signed certs) could have been passed at
the SIP/SDP layer during the establishment of the session in a secure manner
that neither SIP relays nor MSRP relays would be able to thwart. [As an
aside, of course, S/MIME can also provide the necessary end-to-end
authentication through relays, to speak to your second point in
recommendation #4 below.]

But this shows the core issue I have with mandating relay use for its
security properties. You've introduced another entity could be
eavesdropping, and thus you have to use S/MIME to send messages through
relays if you want real security. But if you use S/MIME, you already have
the end-to-end security that you need - you can just use S/MIME between the
endpoints without relays. So what is the security property we really gain
from relays?

These gradations of trust are much more important when considering MSRP
relays than they are in considering SIP relays. You are only required to
trust a SIP proxy server with one piece of salient information, really -
that you want to have a session with a particular someone. You trust the SIP
proxy with the identity of that particular someone because you have to -
that SIP proxy is the only entity that knows how to reach that particular
someone. Once the SIP proxy has connected you to that someone, its job is to
get outta your way and let you chat. The only reason I need to trust an MSRP
proxy is so that it can get me around a network-layer discontinuity (or to
fulfill some operator policy corner-case). I don't want or need to trust it
with the contents of my messages; it shouldn't need to have to inspect them
to provide me network-layer connectivity with someone. If I don't need that
NAT traversal job done, I don't want to run the risk or the cost (in latency
or operational expense) of dealing with an MSRP relay at all. If the
end-to-end model hasn't already been broken for some other reason (by a NAT,
or something), I don't want to break it unnecessarily.

I think the practice of using S/MIME in the absence of relays is described
pretty well already in the fifth and sixth paragraphs of the Security
Considerations of the MSRP spec. The threat model of relays is described in
the MSRP relay draft, in Section 8.3. 

Does that seem reasonable?

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Robert Sparks [mailto:RjS@xten.com]
> Sent: Wednesday, September 08, 2004 8:21 AM
> To: simple@ietf.org
> Cc: Ted Hardie; hartmans@mit.edu
> Subject: [Simple] Security Review of
> draft-ietf-simple-message-sessions-07
> 
> 
[snip]
> >>
> >> 4)  Reconsider the decision to split relay functionality 
> into its own
> >>     draft.  I agree that for many configurations it is 
> impossible to
> >>     get good transport security without relay operation.  
> Rather than
> >>     relaxing the requirement for transport security, I believe that
> >>     the working group should come to the conclusion that 
> relay mode is
> >>     an integral part of the protocol.   Also, the current 
> split does
> >>     not work well because authentication to a relay is 
> different than
> >>     authentication to a peer.  In authenticating to a 
> relay, you are
> >>     authenticating to a specific service running on a 
> host.  However
> >>     when you authenticate to a peer, you are authenticating to a
> >>     specific IM principal.  The current draft does not 
> properly handle
> >>     this distinction  and I don't think a split proposal 
> can easily do
> >>     so.  Note that I have not evaluated the security of the relay
> >>     proposal, but I have considered what security is possible to
> >>     achieve given solutions to the relay problem.
>  

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Subject: [Simple] MSRP Security Review Recommendation 7
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Hi,

In Sam's security review (Thanks Sam!), his recommendation 7 was, and I 
paraphrase, that we should discuss what it means to have messages 
attributed to the same sender that arrive via different paths that may 
have different security properties. I interpret that to mean that the 
messages arrive on different MSRP sessions.

So, here is some proposed text for discussion:

-------------------------------------------------------------
It is possible that a recipient might receive messages that are 
attributed to the same sender via different MSRP sessions. For example, 
Alice might be in a conversation with Bob via an MSRP session over a  
TLS protected channel. Alice might then receive a different message 
from Bob over a different session, perhaps with a conference focus that 
asserts Bob's identity in a message/cpim envelope signed by the focus.

MSRP does not in any way prohibit multiple simultaneous sessions 
between the same pair of identities. Nor does it prohibit an endpoint 
sending a message on behalf of another identity, such as may be the 
case for a conference focus. The recipient's endpoint should determine 
its level of trust of the authenticity of the sender independently for 
each session. The fact that an endpoint trusts the authenticity of the 
sender on any given session should not affect the level of trust it 
assigns for apparently the same sender on a different session.


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From simple-bounces@ietf.org  Thu Sep  9 22:23:21 2004
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Subject: [Simple] MSRP Security Review: RFC3862 Application Considerations
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Hi,

Sam's security review pointed out that we had not handled the 
application considerations required by RFC3862 on all applications 
using the message/cpim MIME type. Here is a first cut at those 
considerations for discussion (quoted text from RFC3862)

>   Applications using this specification must also specify:
>
>    o  all headers that must be recognized by implementations of the
>       application

All MSRP endpoints MUST recognize the From, To, DateTime, and Require 
headers as defined in RFC3862. Such applications SHOULD recognize the 
CC header, and MAY recognize the Subject header. Any MSRP application 
that recognizes any message/cpim header MUST understand the NS (name 
space) header.

>
>    o  any headers that must be present in all messages created by that
>       application.

All message/cpim body parts send by an MSRP endpoint MUST include the 
 From and To headers. If the message/cpim body part is protected using 
S/MIME, then it SHOULD also include the DateTime header.

>
>    o  any headers that may appear more than once in a message, and how
>       they are to be interpreted (e.g., how to interpret multiple
>       'Subject:' headers with different language parameter values).
>

Message/cpim headers defined in RFC 3862, with the exception of the NS 
header, MUST NOT occur more than once in a given message/cpim body part 
in an MSRP message. The Require header MAY include multiple values. The 
NS header MAY occur zero or more times, depending on how many name 
spaces are being referenced.

Extension headers MAY occur more than once, depending on the definition 
of such headers.

>    o  Security mechanisms and crytography schemes to be used with the
>       application, including any mandatory-to-implement security
>       provisions.

[Ben: I _think_ we are covering this in general--do we need to 
explicitly relate such mechanisms to message/cpim?]


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Subject: RE: [Simple] MSRP Security Review Recommendation 7
Date: Fri, 10 Sep 2004 10:53:28 +0300
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Sounds good, although I would not use the term "focus" (unless you want =
to start referencing conferencing documents in sipping).

Regards,
Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Ben Campbell
> Sent: 10.September.2004 04:39
> To: Simple WG
> Subject: [Simple] MSRP Security Review Recommendation 7
>=20
>=20
> Hi,
>=20
> In Sam's security review (Thanks Sam!), his recommendation 7=20
> was, and I=20
> paraphrase, that we should discuss what it means to have messages=20
> attributed to the same sender that arrive via different paths=20
> that may=20
> have different security properties. I interpret that to mean that the=20
> messages arrive on different MSRP sessions.
>=20
> So, here is some proposed text for discussion:
>=20
> -------------------------------------------------------------
> It is possible that a recipient might receive messages that are=20
> attributed to the same sender via different MSRP sessions.=20
> For example,=20
> Alice might be in a conversation with Bob via an MSRP session over a =20
> TLS protected channel. Alice might then receive a different message=20
> from Bob over a different session, perhaps with a conference=20
> focus that=20
> asserts Bob's identity in a message/cpim envelope signed by the focus.
>=20
> MSRP does not in any way prohibit multiple simultaneous sessions=20
> between the same pair of identities. Nor does it prohibit an endpoint=20
> sending a message on behalf of another identity, such as may be the=20
> case for a conference focus. The recipient's endpoint should=20
> determine=20
> its level of trust of the authenticity of the sender=20
> independently for=20
> each session. The fact that an endpoint trusts the=20
> authenticity of the=20
> sender on any given session should not affect the level of trust it=20
> assigns for apparently the same sender on a different session.
>=20
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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Subject: RE: [Simple] MSRP Security Review: RFC3862 Application Considerations
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> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Ben Campbell
> Sent: 10.September.2004 05:22
> To: Simple WG
> Subject: [Simple] MSRP Security Review: RFC3862 Application
> Considerations
>=20
>=20
> Hi,
>=20
> Sam's security review pointed out that we had not handled the=20
> application considerations required by RFC3862 on all applications=20
> using the message/cpim MIME type. Here is a first cut at those=20
> considerations for discussion (quoted text from RFC3862)
>=20
> >   Applications using this specification must also specify:
> >
> >    o  all headers that must be recognized by implementations of the
> >       application
>=20
> All MSRP endpoints MUST recognize the From, To, DateTime, and Require=20
> headers as defined in RFC3862. Such applications SHOULD recognize the=20
> CC header, and MAY recognize the Subject header. Any MSRP application=20
> that recognizes any message/cpim header MUST understand the NS (name=20
> space) header.
>=20
> >
> >    o  any headers that must be present in all messages=20
> created by that
> >       application.
>=20
> All message/cpim body parts send by an MSRP endpoint MUST include the=20
>  From and To headers. If the message/cpim body part is=20
> protected using=20
> S/MIME, then it SHOULD also include the DateTime header.
>=20
> >
> >    o  any headers that may appear more than once in a=20
> message, and how
> >       they are to be interpreted (e.g., how to interpret multiple
> >       'Subject:' headers with different language parameter values).
> >
>=20
> Message/cpim headers defined in RFC 3862, with the exception=20
> of the NS=20
> header, MUST NOT occur more than once in a given message/cpim=20
> body part=20
> in an MSRP message. The Require header MAY include multiple=20
> values. The=20
> NS header MAY occur zero or more times, depending on how many name=20
> spaces are being referenced.
>=20
> Extension headers MAY occur more than once, depending on the=20
> definition=20
> of such headers.

The To-header field should not be limited to 1, nor should the CC field. =
This could be used in the future for, what is commonly referred to as, =
whispering or private messages within a conference.

Also, the subject header might appear multiple times. I would say we =
limit the from-header to one and no limit on the rest.

Regards,
Hisham

>=20
> >    o  Security mechanisms and crytography schemes to be=20
> used with the
> >       application, including any mandatory-to-implement security
> >       provisions.
>=20
> [Ben: I _think_ we are covering this in general--do we need to=20
> explicitly relate such mechanisms to message/cpim?]
>=20
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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From simple-bounces@ietf.org  Fri Sep 10 09:24:16 2004
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From: Ben Campbell <ben@nostrum.com>
Subject: Re: [Simple] MSRP Security Review: RFC3862 Application Considerations
Date: Fri, 10 Sep 2004 08:18:11 -0500
To: <hisham.khartabil@nokia.com>
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On Sep 10, 2004, at 3:08 AM, <hisham.khartabil@nokia.com> wrote:

[...]

>>>    o  any headers that may appear more than once in a
>> message, and how
>>>       they are to be interpreted (e.g., how to interpret multiple
>>>       'Subject:' headers with different language parameter values).
>>>
>>
>> Message/cpim headers defined in RFC 3862, with the exception
>> of the NS
>> header, MUST NOT occur more than once in a given message/cpim
>> body part
>> in an MSRP message. The Require header MAY include multiple
>> values. The
>> NS header MAY occur zero or more times, depending on how many name
>> spaces are being referenced.
>>
>> Extension headers MAY occur more than once, depending on the
>> definition
>> of such headers.
>
> The To-header field should not be limited to 1, nor should the CC 
> field. This could be used in the future for, what is commonly referred 
> to as, whispering or private messages within a conference.
>
> Also, the subject header might appear multiple times. I would say we 
> limit the from-header to one and no limit on the rest.
>

Good point on multiple To and CC values. I'm not sure how CC helps for 
whisper, I think you are looking for something like BCC, which is more 
of a function than a header. Nevertheless, I can see needing multiple 
CC values.


> Regards,
> Hisham
>
>>
>>>    o  Security mechanisms and crytography schemes to be
>> used with the
>>>       application, including any mandatory-to-implement security
>>>       provisions.
>>
>> [Ben: I _think_ we are covering this in general--do we need to
>> explicitly relate such mechanisms to message/cpim?]
>>
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>>


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From simple-bounces@ietf.org  Fri Sep 10 09:37:58 2004
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: hisham.khartabil@nokia.com
Subject: Re: [Simple] MSRP Security Review: RFC3862 Application Considerations
References: <5816828233DEFA41807A6CFDFDF2343C08B5A1@esebe056.ntc.nokia.com>
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Hisham,

I agree with most of your comments, but I question the logic of multiple 
Subject headers. What does this mean? Is the actual subject the 
catenation of the contents of all the Subject headers? Should they have 
<CR><LF> between then when displayed? Or are they to be treated as 
alternatives, such as the same subject represented in different languages?

I believe 3261 only allows one Subject header. I don't know about email, 
but by common usage I don't think it would work well if there was more 
than one.

	Paul

hisham.khartabil@nokia.com wrote:
> 
>>-----Original Message-----
>>From: simple-bounces@ietf.org 
>>[mailto:simple-bounces@ietf.org]On Behalf
>>Of ext Ben Campbell
>>Sent: 10.September.2004 05:22
>>To: Simple WG
>>Subject: [Simple] MSRP Security Review: RFC3862 Application
>>Considerations
>>
>>
>>Hi,
>>
>>Sam's security review pointed out that we had not handled the 
>>application considerations required by RFC3862 on all applications 
>>using the message/cpim MIME type. Here is a first cut at those 
>>considerations for discussion (quoted text from RFC3862)
>>
>>
>>>  Applications using this specification must also specify:
>>>
>>>   o  all headers that must be recognized by implementations of the
>>>      application
>>
>>All MSRP endpoints MUST recognize the From, To, DateTime, and Require 
>>headers as defined in RFC3862. Such applications SHOULD recognize the 
>>CC header, and MAY recognize the Subject header. Any MSRP application 
>>that recognizes any message/cpim header MUST understand the NS (name 
>>space) header.
>>
>>
>>>   o  any headers that must be present in all messages 
>>
>>created by that
>>
>>>      application.
>>
>>All message/cpim body parts send by an MSRP endpoint MUST include the 
>> From and To headers. If the message/cpim body part is 
>>protected using 
>>S/MIME, then it SHOULD also include the DateTime header.
>>
>>
>>>   o  any headers that may appear more than once in a 
>>
>>message, and how
>>
>>>      they are to be interpreted (e.g., how to interpret multiple
>>>      'Subject:' headers with different language parameter values).
>>>
>>
>>Message/cpim headers defined in RFC 3862, with the exception 
>>of the NS 
>>header, MUST NOT occur more than once in a given message/cpim 
>>body part 
>>in an MSRP message. The Require header MAY include multiple 
>>values. The 
>>NS header MAY occur zero or more times, depending on how many name 
>>spaces are being referenced.
>>
>>Extension headers MAY occur more than once, depending on the 
>>definition 
>>of such headers.
> 
> 
> The To-header field should not be limited to 1, nor should the CC field. This could be used in the future for, what is commonly referred to as, whispering or private messages within a conference.
> 
> Also, the subject header might appear multiple times. I would say we limit the from-header to one and no limit on the rest.
> 
> Regards,
> Hisham
> 
> 
>>>   o  Security mechanisms and crytography schemes to be 
>>
>>used with the
>>
>>>      application, including any mandatory-to-implement security
>>>      provisions.
>>
>>[Ben: I _think_ we are covering this in general--do we need to 
>>explicitly relate such mechanisms to message/cpim?]
>>
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>>
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 


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From simple-bounces@ietf.org  Fri Sep 10 11:11:25 2004
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From: Ben Campbell <ben@nostrum.com>
Subject: Re: [Simple] MSRP Security Review: RFC3862 Application Considerations
Date: Fri, 10 Sep 2004 09:53:07 -0500
To: Paul Kyzivat <pkyzivat@cisco.com>
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Oops, in my hurry to reply about To and CC, I missed Subject. I also 
don't know what it means to have multiple subjects. The application 
considerations require us to document what it means to have more than 
one of any particular header that we allow multiples for. Hisham, did 
you have an application in mind?

On Sep 10, 2004, at 8:25 AM, Paul Kyzivat wrote:

> Hisham,
>
> I agree with most of your comments, but I question the logic of 
> multiple Subject headers. What does this mean? Is the actual subject 
> the catenation of the contents of all the Subject headers? Should they 
> have <CR><LF> between then when displayed? Or are they to be treated 
> as alternatives, such as the same subject represented in different 
> languages?
>
> I believe 3261 only allows one Subject header. I don't know about 
> email, but by common usage I don't think it would work well if there 
> was more than one.
>
> 	Paul
>
> hisham.khartabil@nokia.com wrote:
>>> -----Original Message-----
>>> From: simple-bounces@ietf.org [mailto:simple-bounces@ietf.org]On 
>>> Behalf
>>> Of ext Ben Campbell
>>> Sent: 10.September.2004 05:22
>>> To: Simple WG
>>> Subject: [Simple] MSRP Security Review: RFC3862 Application
>>> Considerations
>>>
>>>
>>> Hi,
>>>
>>> Sam's security review pointed out that we had not handled the 
>>> application considerations required by RFC3862 on all applications 
>>> using the message/cpim MIME type. Here is a first cut at those 
>>> considerations for discussion (quoted text from RFC3862)
>>>
>>>
>>>>  Applications using this specification must also specify:
>>>>
>>>>   o  all headers that must be recognized by implementations of the
>>>>      application
>>>
>>> All MSRP endpoints MUST recognize the From, To, DateTime, and 
>>> Require headers as defined in RFC3862. Such applications SHOULD 
>>> recognize the CC header, and MAY recognize the Subject header. Any 
>>> MSRP application that recognizes any message/cpim header MUST 
>>> understand the NS (name space) header.
>>>
>>>
>>>>   o  any headers that must be present in all messages
>>>
>>> created by that
>>>
>>>>      application.
>>>
>>> All message/cpim body parts send by an MSRP endpoint MUST include 
>>> the From and To headers. If the message/cpim body part is protected 
>>> using S/MIME, then it SHOULD also include the DateTime header.
>>>
>>>
>>>>   o  any headers that may appear more than once in a
>>>
>>> message, and how
>>>
>>>>      they are to be interpreted (e.g., how to interpret multiple
>>>>      'Subject:' headers with different language parameter values).
>>>>
>>>
>>> Message/cpim headers defined in RFC 3862, with the exception of the 
>>> NS header, MUST NOT occur more than once in a given message/cpim 
>>> body part in an MSRP message. The Require header MAY include 
>>> multiple values. The NS header MAY occur zero or more times, 
>>> depending on how many name spaces are being referenced.
>>>
>>> Extension headers MAY occur more than once, depending on the 
>>> definition of such headers.
>> The To-header field should not be limited to 1, nor should the CC 
>> field. This could be used in the future for, what is commonly 
>> referred to as, whispering or private messages within a conference.
>> Also, the subject header might appear multiple times. I would say we 
>> limit the from-header to one and no limit on the rest.
>> Regards,
>> Hisham
>>>>   o  Security mechanisms and crytography schemes to be
>>>
>>> used with the
>>>
>>>>      application, including any mandatory-to-implement security
>>>>      provisions.
>>>
>>> [Ben: I _think_ we are covering this in general--do we need to 
>>> explicitly relate such mechanisms to message/cpim?]
>>>
>>>
>>> _______________________________________________
>>> Simple mailing list
>>> Simple@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/simple
>>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple


_______________________________________________
Simple mailing list
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Subject: RE: [Simple] MSRP Security Review: RFC3862 Application Considerations
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> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 10.September.2004 16:26
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: ben@nostrum.com; simple@ietf.org
> Subject: Re: [Simple] MSRP Security Review: RFC3862 Application
> Considerations
>=20
>=20
> Hisham,
>=20
> I agree with most of your comments, but I question the logic=20
> of multiple=20
> Subject headers. What does this mean? Is the actual subject the=20
> catenation of the contents of all the Subject headers? Should=20
> they have=20
> <CR><LF> between then when displayed? Or are they to be treated as=20
> alternatives, such as the same subject represented in=20
> different languages?

That's what I had in mind.

/Hisham

>=20
> I believe 3261 only allows one Subject header. I don't know=20
> about email,=20
> but by common usage I don't think it would work well if there=20
> was more=20
> than one.
>=20
> 	Paul
>=20
> hisham.khartabil@nokia.com wrote:
> >=20
> >>-----Original Message-----
> >>From: simple-bounces@ietf.org=20
> >>[mailto:simple-bounces@ietf.org]On Behalf
> >>Of ext Ben Campbell
> >>Sent: 10.September.2004 05:22
> >>To: Simple WG
> >>Subject: [Simple] MSRP Security Review: RFC3862 Application
> >>Considerations
> >>
> >>
> >>Hi,
> >>
> >>Sam's security review pointed out that we had not handled the=20
> >>application considerations required by RFC3862 on all applications=20
> >>using the message/cpim MIME type. Here is a first cut at those=20
> >>considerations for discussion (quoted text from RFC3862)
> >>
> >>
> >>>  Applications using this specification must also specify:
> >>>
> >>>   o  all headers that must be recognized by implementations of the
> >>>      application
> >>
> >>All MSRP endpoints MUST recognize the From, To, DateTime,=20
> and Require=20
> >>headers as defined in RFC3862. Such applications SHOULD=20
> recognize the=20
> >>CC header, and MAY recognize the Subject header. Any MSRP=20
> application=20
> >>that recognizes any message/cpim header MUST understand the=20
> NS (name=20
> >>space) header.
> >>
> >>
> >>>   o  any headers that must be present in all messages=20
> >>
> >>created by that
> >>
> >>>      application.
> >>
> >>All message/cpim body parts send by an MSRP endpoint MUST=20
> include the=20
> >> From and To headers. If the message/cpim body part is=20
> >>protected using=20
> >>S/MIME, then it SHOULD also include the DateTime header.
> >>
> >>
> >>>   o  any headers that may appear more than once in a=20
> >>
> >>message, and how
> >>
> >>>      they are to be interpreted (e.g., how to interpret multiple
> >>>      'Subject:' headers with different language parameter values).
> >>>
> >>
> >>Message/cpim headers defined in RFC 3862, with the exception=20
> >>of the NS=20
> >>header, MUST NOT occur more than once in a given message/cpim=20
> >>body part=20
> >>in an MSRP message. The Require header MAY include multiple=20
> >>values. The=20
> >>NS header MAY occur zero or more times, depending on how many name=20
> >>spaces are being referenced.
> >>
> >>Extension headers MAY occur more than once, depending on the=20
> >>definition=20
> >>of such headers.
> >=20
> >=20
> > The To-header field should not be limited to 1, nor should=20
> the CC field. This could be used in the future for, what is=20
> commonly referred to as, whispering or private messages=20
> within a conference.
> >=20
> > Also, the subject header might appear multiple times. I=20
> would say we limit the from-header to one and no limit on the rest.
> >=20
> > Regards,
> > Hisham
> >=20
> >=20
> >>>   o  Security mechanisms and crytography schemes to be=20
> >>
> >>used with the
> >>
> >>>      application, including any mandatory-to-implement security
> >>>      provisions.
> >>
> >>[Ben: I _think_ we are covering this in general--do we need to=20
> >>explicitly relate such mechanisms to message/cpim?]
> >>
> >>
> >>_______________________________________________
> >>Simple mailing list
> >>Simple@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/simple
> >>
> >=20
> >=20
> > _______________________________________________
> > Simple mailing list
> > Simple@ietf.org
> > https://www1.ietf.org/mailman/listinfo/simple
> >=20
>=20
>=20

_______________________________________________
Simple mailing list
Simple@ietf.org
https://www1.ietf.org/mailman/listinfo/simple


From simple-bounces@ietf.org  Mon Sep 13 07:19:33 2004
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From: "Alexeitsev, D" <D.Alexeitsev@t-com.net>
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Hi All,

I would like to discuss the issue of resource lists URI starting with the introduction.

Resource lists draft does not give much guidelines about the creation of URI for the resource lists.

It is only says that it is a SIP URI and from the given examples one can assume that the URIs are taken from the same namespace as URIs for the "real" resources that actually provide presence information.

I can see the idea behind looking at the resource lists from the namespace perspective just as at any other resource that one can subscribe to receive presence information.

However such definition complicates the assignment of the URI to the "real" resources (subscribers) as one must check if the URI that subscriber has picked up (e.g. Web-Provisioning) is not only already taken by another subscriber, but also if it is not already somewhere taken for the resource list.

Additionally there is no automatic provisioning at the UA, as subscribers will have to fill just another form to point to the resource list URI.

A proposal would be to use something like user=resource-list URI parameter to separate this namespaces. Provisioning of the resource list URI in the UA could be done automatically by the UA, as it would be the same as the subscribers own URI with the the user=resource-list parameter. 

Surly extra resource lists URIs can be assign to the subscriber, but that would be a specific add-on for which subscribers will have to take the pain to manually provision them in to their UAs.    

Greetings,
Denis Alexeitsev

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From simple-bounces@ietf.org  Mon Sep 13 08:56:14 2004
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Denis,

I'm sure there are a multitude of ways to do this, and I'm not convinced 
it needs to be a matter for IETF standardization.

Personally, I have been thinking that a resource list server would 
normally have a domain name distinct from that normally used for "user" 
AORs.

For instance:

- User AORs:
   sip:alice@acme.com

- Resource list AORs:
   sip:alice-friends@lists.acme.com
   sip:retailers@lists.acme.com

	Paul

Alexeitsev, D wrote:
> Hi All,
> 
> I would like to discuss the issue of resource lists URI starting with the introduction.
> 
> Resource lists draft does not give much guidelines about the creation of URI for the resource lists.
> 
> It is only says that it is a SIP URI and from the given examples one can assume that the URIs are taken from the same namespace as URIs for the "real" resources that actually provide presence information.
> 
> I can see the idea behind looking at the resource lists from the namespace perspective just as at any other resource that one can subscribe to receive presence information.
> 
> However such definition complicates the assignment of the URI to the "real" resources (subscribers) as one must check if the URI that subscriber has picked up (e.g. Web-Provisioning) is not only already taken by another subscriber, but also if it is not already somewhere taken for the resource list.
> 
> Additionally there is no automatic provisioning at the UA, as subscribers will have to fill just another form to point to the resource list URI.
> 
> A proposal would be to use something like user=resource-list URI parameter to separate this namespaces. Provisioning of the resource list URI in the UA could be done automatically by the UA, as it would be the same as the subscribers own URI with the the user=resource-list parameter. 
> 
> Surly extra resource lists URIs can be assign to the subscriber, but that would be a specific add-on for which subscribers will have to take the pain to manually provision them in to their UAs.    
> 
> Greetings,
> Denis Alexeitsev
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 


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From: "Alexeitsev, D" <D.Alexeitsev@t-com.net>
To: pkyzivat@cisco.com
Subject: Re: [Simple] URI for Resource Lists
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Hello Paul,

Thanks for the reply.

>I'm sure there are a multitude of ways to do this, and I'm not convinced 
>it needs to be a matter for IETF standardization.

Well in my opinion if there are multitude of ways to do something, then it is definitely a topic for standardisation. In the case of resource lists I think even an information description of a common way for RLS naming would be helpful. 

Using a domain name differentiation is a nice one, are there any other views on the list? 


>Personally, I have been thinking that a resource list server would 
>normally have a domain name distinct from that normally used for "user" 
>AORs.

>For instance:
>
>- User AORs:
>   sip:alice@acme.com
>
>- Resource list AORs:
>   sip:alice-friends@lists.acme.com
>   sip:retailers@lists.acme.com


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Subject: [Simple] FW: CPCP Issue 37: Generic privileges solution
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The draft can be found at:
http://www.imsbook.com/hisham/draft-ietf-xcon-conference-policy-privilege=
s-00.txt

SIMPLE WG comments are being sorted on this issue presented below.

Thanks,
Hisham

> -----Original Message-----
> From: Khartabil Hisham (Nokia-TP-MSW/Helsinki)=20
> Sent: 13.September.2004 16:46
> To: 'xcon@ietf.org'
> Subject: CPCP Issue 37: Generic privileges solution
>=20
>=20
> This issue affects=20
> draft-ietf-xcon-conference-policy-privileges-00.txt.
>=20
> The schema in this draft is generic and can be used for any=20
> XML document that carries user information and resides on a=20
> server. It is useful especially for XCAP.
>=20
> I have been toying with the idea of extracting the schema=20
> into a separate draft and submitting that in SIMPLE. XCON can=20
> then use that draft and customise it for conferencing needs.=20
> By customising it, I mean defining the actions and=20
> transformations that grant the conferencing specific=20
> privileges to users.
>=20
> This solution uses common policy as well where the conditions=20
> carry the document URI and the identity of users and the=20
> actions carry the privileges (read/modify certain parts of=20
> the conference policy).
>=20
> BTW, the draft defines a sematic approach as agreed in San Diego.
>=20
> So, what do people think of that?
>=20
> I'll forward this email to SIMPLE for comments.
>=20
> Regards,
> Hisham
>=20

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From simple-bounces@ietf.org  Tue Sep 14 03:03:38 2004
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To: sip-implementors-request@cs.columbia.edu, simple@ietf.org
From: Robert Sparks <RjS@xten.com>
Date: Tue, 14 Sep 2004 01:49:29 -0500
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Subject: [Simple] Registration for SIMPLEt 3 is open
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SIMPLEt 3 will be held in Redmond, Washington November 1-3.
This SIMPLEt is hosted by Microsoft, and there is no registration fee.
All you need to attend is your own SIMPLE implementation to test.

The event will be held in the Microsoft Platform Lab.
See http://www.microsoftlab.net/ for information on how to get there 
and where to stay.

If you're planning to attend, please register today by sending email to 
simplet3@microsoft.com.
The registration deadline is October 25.

You can find more information about the SIPITs and SIMPLEts at
http://www.sipit.net/.

See you in Redmond!

RjS


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From simple-bounces@ietf.org  Tue Sep 14 11:16:39 2004
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Subject: [Simple] Strength of XML validity requirements in SIMPLE documents
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Folks -

We have several drafts that define XML documents. These drafts
all say something along the lines of

"The simple-filter is an XML document [8] that MUST be well-formed and
    SHOULD be valid."

Why are we specifying SHOULD be valid instead of MUST? I'm asking
this for both simple-filter and for the SIMPLE documents in general.

RjS


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From simple-bounces@ietf.org  Tue Sep 14 14:30:48 2004
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Hmm, its a good question. I think that the vast majority of XML docs
produced in the SIP/SIPPING/SIMPLE groups have the same words, and to be 
honest, I had not
given a ton of thought to MUST vs. SHOULD for valid. I can't really
think of a reason why it would ever be the right thing to produce a
document that is not valid, particularly if the recipient does schema
validation. Certainly I think the actual validation of documents should
never be MUST, excepting rare cases like xcap when its a key part of the
protocol. But, thats a different issue.


-Jonathan R.

Robert Sparks wrote:

> Folks -
> 
> We have several drafts that define XML documents. These drafts
> all say something along the lines of
> 
> "The simple-filter is an XML document [8] that MUST be well-formed and
>    SHOULD be valid."
> 
> Why are we specifying SHOULD be valid instead of MUST? I'm asking
> this for both simple-filter and for the SIMPLE documents in general.
> 
> RjS
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Pekka Pessi <Pekka.Pessi@nokia.com>
Date: Wed, 15 Sep 2004 18:53:04 +0300
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Hello all SUBSCRIBE/NOTIFY lawyers,

The RFC 3265 says that the NOTIFY sent in confirmation of a
subscription MAY contain and empty or neutral body if the resource
has no meaningful state. RFC 3856 (and partial-notify) also mention
that a NOTIFY MAY be sent without a body. 

Now, what a subscriber does when he receives a NOTIFY with
Content-Length: 0 in the middle of a subscription? Can he conclude
that the event state is not meaningful anymore and ditch the state
from previous NOTIFY? Or can he do that only if there is
a Content-Type header in the NOTIFY?

--Pekka

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Pekka Pessi <Pekka.Pessi@nokia.com>
Subject: Re: [Simple] NOTIFY without message-body
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If the Content-TYpe is present, and the length is zero, it means that 
the body is of that type but has no information in it. I'll note that 
this is probably illegal, since a PIDF document is supposed to be 
valid/well-formed, and null content doesnt have that property.

If the Content-Type is absent and the length is zero, it means that 
there is no body, and the state of the watched resource is unchanged.

-Jonathan R.

Pekka Pessi wrote:

> Hello all SUBSCRIBE/NOTIFY lawyers,
> 
> The RFC 3265 says that the NOTIFY sent in confirmation of a
> subscription MAY contain and empty or neutral body if the resource
> has no meaningful state. RFC 3856 (and partial-notify) also mention
> that a NOTIFY MAY be sent without a body. 
> 
> Now, what a subscriber does when he receives a NOTIFY with
> Content-Length: 0 in the middle of a subscription? Can he conclude
> that the event state is not meaningful anymore and ditch the state
> from previous NOTIFY? Or can he do that only if there is
> a Content-Type header in the NOTIFY?
> 
> --Pekka
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Simple mailing list
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https://www1.ietf.org/mailman/listinfo/simple


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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
In-Reply-To: <414735F4.1070302@dynamicsoft.com> (Jonathan Rosenberg's
	message of "Tue, 14 Sep 2004 14:18:28 -0400")
References: <EDB722BE-065D-11D9-A64D-000D93326732@xten.com>
	<414735F4.1070302@dynamicsoft.com>
Date: Wed, 15 Sep 2004 20:04:11 +0300
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Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>Hmm, its a good question. I think that the vast majority of XML
>docs produced in the SIP/SIPPING/SIMPLE groups have the same
>words, and to be honest, I had not given a ton of thought to MUST
>vs. SHOULD for valid. I can't really think of a reason why it
>would ever be the right thing to produce a document that is not
>valid, particularly if the recipient does schema validation.

The presence documents may contain all kind of extensions, which
may be only meaningful for a single application using the presence
service. The presence servers or any intermediate nodes (not to
speak about any innocent watcher) can't really know if the
documents are valid unless they know the schema for all of
those extensions.

Like, if composer puts two tuples from two different sources to
the combined document in the presence server. Both the extensions
use ID attributes, but how could the composer know which
attributes have xs:ID type? The following example may be valid, or
then not, but who is to say without a schema?

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
     xmlns:x="http://example.com/ex/ten/sion"
     entity="pres:john.doe@example.com">
  <tuple id="id1"><status><basic>open</basic></status></tuple>
  <x:example-extension id="id1" />
</presence>

--Pekka

>Robert Sparks wrote:

>> Folks -
>> We have several drafts that define XML documents. These drafts
>> all say something along the lines of
>> "The simple-filter is an XML document [8] that MUST be well-formed and
>>    SHOULD be valid."
>> Why are we specifying SHOULD be valid instead of MUST? I'm asking
>> this for both simple-filter and for the SIMPLE documents in general.
>> RjS
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple

_______________________________________________
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From simple-bounces@ietf.org  Wed Sep 15 14:02:29 2004
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From: Robert Sparks <RjS@xten.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Wed, 15 Sep 2004 12:39:35 -0500
To: Pekka Pessi <Pekka.Pessi@nokia.com>
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This makes sense. For work going forward, we should have text
in the document definition drafts that call out that this _kind_ of 
thing
can happen.

In the drafts that use the document, though, we might be able to be
more strict.

For instance, consider filter-format and filter-funct.
Would it be helpful for  filter format to say SHOULD be valid,
but filter-funct say that a filter appearing in the body of a
SUBSCRIBE MUST be valid?

RjS


On Sep 15, 2004, at 12:04 PM, Pekka Pessi wrote:

> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>> Hmm, its a good question. I think that the vast majority of XML
>> docs produced in the SIP/SIPPING/SIMPLE groups have the same
>> words, and to be honest, I had not given a ton of thought to MUST
>> vs. SHOULD for valid. I can't really think of a reason why it
>> would ever be the right thing to produce a document that is not
>> valid, particularly if the recipient does schema validation.
>
> The presence documents may contain all kind of extensions, which
> may be only meaningful for a single application using the presence
> service. The presence servers or any intermediate nodes (not to
> speak about any innocent watcher) can't really know if the
> documents are valid unless they know the schema for all of
> those extensions.
>
> Like, if composer puts two tuples from two different sources to
> the combined document in the presence server. Both the extensions
> use ID attributes, but how could the composer know which
> attributes have xs:ID type? The following example may be valid, or
> then not, but who is to say without a schema?
>
> <?xml version="1.0" encoding="UTF-8"?>
> <presence xmlns="urn:ietf:params:xml:ns:pidf"
>      xmlns:x="http://example.com/ex/ten/sion"
>      entity="pres:john.doe@example.com">
>   <tuple id="id1"><status><basic>open</basic></status></tuple>
>   <x:example-extension id="id1" />
> </presence>
>
> --Pekka
>
>> Robert Sparks wrote:
>
>>> Folks -
>>> We have several drafts that define XML documents. These drafts
>>> all say something along the lines of
>>> "The simple-filter is an XML document [8] that MUST be well-formed 
>>> and
>>>    SHOULD be valid."
>>> Why are we specifying SHOULD be valid instead of MUST? I'm asking
>>> this for both simple-filter and for the SIMPLE documents in general.
>>> RjS
>>> _______________________________________________
>>> Simple mailing list
>>> Simple@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/simple
>



_______________________________________________
Simple mailing list
Simple@ietf.org
https://www1.ietf.org/mailman/listinfo/simple


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Date: Wed, 15 Sep 2004 10:59:38 -0700 (PDT)
From: Fatih "Eyüp" NAR <fenar@yahoo.com>
Subject: Re: [Simple] NOTIFY without message-body
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Pekka Pessi <Pekka.Pessi@nokia.com>
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does this comment only applicable to Notify or for all
content embedded SIP-Requests? (ex:Register)

ex:(as far as I saw/understood)
if REGISTER with content/xml with content.length=0 is
overriding the previously received/sent Register with
content/xml with full body(i.e content.length>0) ,
which is legal .

--- Jonathan Rosenberg <jdrosen@dynamicsoft.com>
wrote:

> If the Content-TYpe is present, and the length is
> zero, it means that 
> the body is of that type but has no information in
> it. I'll note that 
> this is probably illegal, since a PIDF document is
> supposed to be 
> valid/well-formed, and null content doesnt have that
> property.
> 
> If the Content-Type is absent and the length is
> zero, it means that 
> there is no body, and the state of the watched
> resource is unchanged.
> 
> -Jonathan R.
> 
> Pekka Pessi wrote:
> 
> > Hello all SUBSCRIBE/NOTIFY lawyers,
> > 
> > The RFC 3265 says that the NOTIFY sent in
> confirmation of a
> > subscription MAY contain and empty or neutral body
> if the resource
> > has no meaningful state. RFC 3856 (and
> partial-notify) also mention
> > that a NOTIFY MAY be sent without a body. 
> > 
> > Now, what a subscriber does when he receives a
> NOTIFY with
> > Content-Length: 0 in the middle of a subscription?
> Can he conclude
> > that the event state is not meaningful anymore and
> ditch the state
> > from previous NOTIFY? Or can he do that only if
> there is
> > a Content-Type header in the NOTIFY?
> > 
> > --Pekka
> > 
> > _______________________________________________
> > Simple mailing list
> > Simple@ietf.org
> > https://www1.ietf.org/mailman/listinfo/simple
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                600
> Lanidex Plaza
> Chief Technology Officer                   
> Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:  
> (973) 952-5050
> http://www.jdrosen.net                      PHONE:
> (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 



	
		
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Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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inline.

Pekka Pessi wrote:

> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> 
>>Hmm, its a good question. I think that the vast majority of XML
>>docs produced in the SIP/SIPPING/SIMPLE groups have the same
>>words, and to be honest, I had not given a ton of thought to MUST
>>vs. SHOULD for valid. I can't really think of a reason why it
>>would ever be the right thing to produce a document that is not
>>valid, particularly if the recipient does schema validation.
> 
> 
> The presence documents may contain all kind of extensions, which
> may be only meaningful for a single application using the presence
> service. The presence servers or any intermediate nodes (not to
> speak about any innocent watcher) can't really know if the
> documents are valid unless they know the schema for all of
> those extensions.
> 
> Like, if composer puts two tuples from two different sources to
> the combined document in the presence server. Both the extensions
> use ID attributes, but how could the composer know which
> attributes have xs:ID type? The following example may be valid, or
> then not, but who is to say without a schema?

It depends on how you want to define "valid". If a document has some 
extensions (say, rpid ones), such a document is still valid according to 
just the pidf schema, since it contains well-formed elements from 
another namespace in a place allowed by the schema.

So, when we say "SHOULD be valid", do we mean that all elements in the 
document should be valid according to their defined schemas, or just 
that the document is valid according to the pidf schema?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From simple-bounces@ietf.org  Thu Sep 16 00:02:08 2004
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To: =?ISO-8859-1?Q?=22Fatih_=5C=22Ey=FCp=5C=22_NAR=22?= <fenar@yahoo.com>
Subject: Re: [Simple] NOTIFY without message-body
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It would be applicable to other requests where there is a=20
content/non-content choice that is meaningful. REGISTER is not a good=20
example since we don't actually define the meaning of content there.=20
UPDATE is a better example. It can have an SDP offer (CT:=20
application/sdp, CL: non-zero) or it can have none (no CT, CL: 0). There=20
also, a CT of application/sdp and a CL of zero is not valid, since a=20
null document would not be valid SDP.

MESSAGE is another example. There, there can be a text message (CT:=20
text/plain, CL: non-zero), no content at all (no CT, CL: 0), or a=20
message with content that is "null" (CT: text/plain, CL: 0).

-Jonathan R.

Fatih Ey=FCp NAR wrote:

> does this comment only applicable to Notify or for all
> content embedded SIP-Requests? (ex:Register)
>=20
> ex:(as far as I saw/understood)
> if REGISTER with content/xml with content.length=3D0 is
> overriding the previously received/sent Register with
> content/xml with full body(i.e content.length>0) ,
> which is legal .
>=20
> --- Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> wrote:
>=20
>=20
>>If the Content-TYpe is present, and the length is
>>zero, it means that=20
>>the body is of that type but has no information in
>>it. I'll note that=20
>>this is probably illegal, since a PIDF document is
>>supposed to be=20
>>valid/well-formed, and null content doesnt have that
>>property.
>>
>>If the Content-Type is absent and the length is
>>zero, it means that=20
>>there is no body, and the state of the watched
>>resource is unchanged.
>>
>>-Jonathan R.
>>
>>Pekka Pessi wrote:
>>
>>
>>>Hello all SUBSCRIBE/NOTIFY lawyers,
>>>
>>>The RFC 3265 says that the NOTIFY sent in
>>
>>confirmation of a
>>
>>>subscription MAY contain and empty or neutral body
>>
>>if the resource
>>
>>>has no meaningful state. RFC 3856 (and
>>
>>partial-notify) also mention
>>
>>>that a NOTIFY MAY be sent without a body.=20
>>>
>>>Now, what a subscriber does when he receives a
>>
>>NOTIFY with
>>
>>>Content-Length: 0 in the middle of a subscription?
>>
>>Can he conclude
>>
>>>that the event state is not meaningful anymore and
>>
>>ditch the state
>>
>>>from previous NOTIFY? Or can he do that only if
>>
>>there is
>>
>>>a Content-Type header in the NOTIFY?
>>>
>>>--Pekka
>>>
>>>_______________________________________________
>>>Simple mailing list
>>>Simple@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/simple
>>>
>>
>>--=20
>>Jonathan D. Rosenberg, Ph.D.                600
>>Lanidex Plaza
>>Chief Technology Officer                  =20
>>Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX: =20
>>(973) 952-5050
>>http://www.jdrosen.net                      PHONE:
>>(973) 952-5000
>>http://www.dynamicsoft.com
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>>
>=20
>=20
>=20
>=20
> =09
> 	=09
> __________________________________
> Do you Yahoo!?
> New and Improved Yahoo! Mail - 100MB free storage!
> http://promotions.yahoo.com/new_mail=20
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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A few comments here.

Firstly, the XCAP resource list usage 
(http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-list-usage-03.txt) 
provides a way for clients to request and/or create URIs for resource 
lists. This specification mandates that the server ensure that the URI 
provided for the resource list is unique within the domain. As you point 
out, this uniqueness requirement spans both other resource list URIs, 
and regular users in that domain. It would also include any other SIP 
resources  - voicemail servers, applications, etc.

I don't think we need to dictate how this gets done. I diagree that, 
just because there is variability, there needs to be standardization. 
Variability requires standardization only when, if one entity does 
something different than another, there are interop problems. I don't 
believe that is the case here. I believe this is squarely in the area of 
*policy*, and that is something we should stay far away from. Just like 
we don't dictate that your SIP URI has to be 
sip:<firstname>.<lastname>@your-domain.com, we should not dictate how 
resource list URIs look like.

Thanks,
Jonathan R.

Alexeitsev, D wrote:

> Hello Paul,
> 
> Thanks for the reply.
> 
> 
>> I'm sure there are a multitude of ways to do this, and I'm not
>> convinced it needs to be a matter for IETF standardization.
> 
> 
> Well in my opinion if there are multitude of ways to do something,
> then it is definitely a topic for standardisation. In the case of
> resource lists I think even an information description of a common
> way for RLS naming would be helpful.
> 
> Using a domain name differentiation is a nice one, are there any
> other views on the list?
> 
> 
> 
>> Personally, I have been thinking that a resource list server would
>>  normally have a domain name distinct from that normally used for
>> "user" AORs.
> 
> 
>> For instance:
>> 
>> - User AORs: sip:alice@acme.com
>> 
>> - Resource list AORs: sip:alice-friends@lists.acme.com 
>> sip:retailers@lists.acme.com
> 
> 
> 
> _______________________________________________ Simple mailing list 
> Simple@ietf.org https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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To: ext Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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ext Jonathan Rosenberg wrote:

> inline.
>
> Pekka Pessi wrote:
>
>> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>>
>>> Hmm, its a good question. I think that the vast majority of XML
>>> docs produced in the SIP/SIPPING/SIMPLE groups have the same
>>> words, and to be honest, I had not given a ton of thought to MUST
>>> vs. SHOULD for valid. I can't really think of a reason why it
>>> would ever be the right thing to produce a document that is not
>>> valid, particularly if the recipient does schema validation.
>>
>>
>>
>> The presence documents may contain all kind of extensions, which
>> may be only meaningful for a single application using the presence
>> service. The presence servers or any intermediate nodes (not to
>> speak about any innocent watcher) can't really know if the
>> documents are valid unless they know the schema for all of
>> those extensions.
>>
>> Like, if composer puts two tuples from two different sources to
>> the combined document in the presence server. Both the extensions
>> use ID attributes, but how could the composer know which
>> attributes have xs:ID type? The following example may be valid, or
>> then not, but who is to say without a schema?
>
>
> It depends on how you want to define "valid". If a document has some 
> extensions (say, rpid ones), such a document is still valid according 
> to just the pidf schema, since it contains well-formed elements from 
> another namespace in a place allowed by the schema.
>
> So, when we say "SHOULD be valid", do we mean that all elements in the 
> document should be valid according to their defined schemas, or just 
> that the document is valid according to the pidf schema?
>
> -Jonathan R.

IMO pidf schema requirement is a "MUST" as content-type 
application/pidf+xml is defined by that schema. How could you use some 
instance document with that content-type if it is not valid ? For 
extensions, instance documents can use schemaLocation attribute, but it 
is only a hint to the validator.
--Jari



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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Simple] NOTIFY without message-body
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
In-Reply-To: <41487517.9070308@dynamicsoft.com> (Jonathan Rosenberg's
	message of "Wed, 15 Sep 2004 13:00:07 -0400")
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	<41487517.9070308@dynamicsoft.com>
Date: Thu, 16 Sep 2004 13:45:58 +0300
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Hi Jonathan,

>If the Content-TYpe is present, and the length is zero, it means that the
>body is of that type but has no information in it. I'll note that this is
>probably illegal, since a PIDF document is supposed to be valid/well-formed,
>and null content doesnt have that property.

>If the Content-Type is absent and the length is zero, it means that there is
>no body, and the state of the watched resource is unchanged.

OK. So, if I understood the specs correctly, the notifier can send
a NOTIFY without body at any time. 

Now, let's assume that the subscriber is out of sync with, for
instance, partial presence notifications. How it can ask the
server to send a NOTIFY with full state? It could terminate and
recreate the subscription, but that potentially involve a large
number of message.

--Pekka

>Pekka Pessi wrote:
>> Hello all SUBSCRIBE/NOTIFY lawyers,
>> The RFC 3265 says that the NOTIFY sent in confirmation of a
>> subscription MAY contain and empty or neutral body if the resource
>> has no meaningful state. RFC 3856 (and partial-notify) also mention
>> that a NOTIFY MAY be sent without a body. Now, what a subscriber does when
>> he receives a NOTIFY with
>> Content-Length: 0 in the middle of a subscription? Can he conclude
>> that the event state is not meaningful anymore and ditch the state
>> from previous NOTIFY? Or can he do that only if there is
>> a Content-Type header in the NOTIFY?
>> --Pekka
>> _______________________________________________
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Pekka Pessi" <Pekka.Pessi@nokia.com>
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Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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Please see below...
----- Original Message ----- 
From: "Pekka Pessi" <Pekka.Pessi@nokia.com>


> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> >Hmm, its a good question. I think that the vast majority of XML
> >docs produced in the SIP/SIPPING/SIMPLE groups have the same
> >words, and to be honest, I had not given a ton of thought to MUST
> >vs. SHOULD for valid. I can't really think of a reason why it
> >would ever be the right thing to produce a document that is not
> >valid, particularly if the recipient does schema validation.
>
> The presence documents may contain all kind of extensions, which
> may be only meaningful for a single application using the presence
> service. The presence servers or any intermediate nodes (not to
> speak about any innocent watcher) can't really know if the
> documents are valid unless they know the schema for all of
> those extensions.
>
> Like, if composer puts two tuples from two different sources to
> the combined document in the presence server. Both the extensions
> use ID attributes, but how could the composer know which
> attributes have xs:ID type? The following example may be valid, or
> then not, but who is to say without a schema?
>
> <?xml version="1.0" encoding="UTF-8"?>
> <presence xmlns="urn:ietf:params:xml:ns:pidf"
>      xmlns:x="http://example.com/ex/ten/sion"
>      entity="pres:john.doe@example.com">
>   <tuple id="id1"><status><basic>open</basic></status></tuple>
>   <x:example-extension id="id1" />
> </presence>
>


I don't know about the specifics of the presence schema, but presumably the
original presence schema specifies <xs:any namespace='##other'/> at some key
points.  Hence the document is valid against the original schema.
(Mumbo-jumbo is valid in the schema sense as long as it is from another
namespace.)  Put another way, a document can be valid in the schema sense,
but still not be semantically correct.

How validity is established for the extensions is another issue.  This comes
down to versioning.  I think this is an area where the group(s) needs to get
a common understanding on how things should be done.  (Schema is actually
weak in this area!  There are ways, but they don't come for free and they
certainly don't happen by accident!)

Pete.
--
=============================================
Pete Cordell
Tech-Know-Ware
pete(at)tech-know-ware.com
http://www.tech-know-ware.com
http://www.xml2cpp.com - Now Available
+44 1473 635863
=============================================


> --Pekka
>
> >Robert Sparks wrote:
... cut ...


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From: "Alexeitsev, D" <D.Alexeitsev@t-com.net>
To: jdrosen@dynamicsoft.com
Subject: AW: [Simple] URI for Resource Lists
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Hello Jonathan

Would then the sip:alice@example.com;user=lists meet the uniqueness requirement if there is already a resource named sip:alice@example.com?

I assume that the example sip:alice@lists.example.com would meet this requirement as it is in the different domain as sip:alice@example.com


>Firstly, the XCAP resource list usage 
>(http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-list-usage-03.txt) 
>provides a way for clients to request and/or create URIs for resource 
>lists. This specification mandates that the server ensure that the URI 
>provided for the resource list is unique within the domain. As you point 
>out, this uniqueness requirement spans both other resource list URIs, 
>and regular users in that domain. It would also include any other SIP 
>resources  - voicemail servers, applications, etc.

Greetings,
Denis Alexeitsev

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Jonathan,

What you suggest is plausible, but I don't recall ever seeing anything=20
written that discusses the meaning of a Content-Type with an empty body.

That logic would imply that there is (conceptually) a special content=20
type for which an empty body is valid and a non-empty body is not, and=20
that this is a default content-type when the body is empty.

I believe I have seen stacks that considered the presence of a content=20
type header invalid if there is no body. At the least there ought to be=20
something that says this is valid usage.

	Paul

Jonathan Rosenberg wrote:
> It would be applicable to other requests where there is a=20
> content/non-content choice that is meaningful. REGISTER is not a good=20
> example since we don't actually define the meaning of content there.=20
> UPDATE is a better example. It can have an SDP offer (CT:=20
> application/sdp, CL: non-zero) or it can have none (no CT, CL: 0). Ther=
e=20
> also, a CT of application/sdp and a CL of zero is not valid, since a=20
> null document would not be valid SDP.
>=20
> MESSAGE is another example. There, there can be a text message (CT:=20
> text/plain, CL: non-zero), no content at all (no CT, CL: 0), or a=20
> message with content that is "null" (CT: text/plain, CL: 0).
>=20
> -Jonathan R.
>=20
> Fatih Ey=FCp NAR wrote:
>=20
>> does this comment only applicable to Notify or for all
>> content embedded SIP-Requests? (ex:Register)
>>
>> ex:(as far as I saw/understood)
>> if REGISTER with content/xml with content.length=3D0 is
>> overriding the previously received/sent Register with
>> content/xml with full body(i.e content.length>0) ,
>> which is legal .
>>
>> --- Jonathan Rosenberg <jdrosen@dynamicsoft.com>
>> wrote:
>>
>>
>>> If the Content-TYpe is present, and the length is
>>> zero, it means that the body is of that type but has no information i=
n
>>> it. I'll note that this is probably illegal, since a PIDF document is
>>> supposed to be valid/well-formed, and null content doesnt have that
>>> property.
>>>
>>> If the Content-Type is absent and the length is
>>> zero, it means that there is no body, and the state of the watched
>>> resource is unchanged.
>>>
>>> -Jonathan R.
>>>
>>> Pekka Pessi wrote:
>>>
>>>
>>>> Hello all SUBSCRIBE/NOTIFY lawyers,
>>>>
>>>> The RFC 3265 says that the NOTIFY sent in
>>>
>>>
>>> confirmation of a
>>>
>>>> subscription MAY contain and empty or neutral body
>>>
>>>
>>> if the resource
>>>
>>>> has no meaningful state. RFC 3856 (and
>>>
>>>
>>> partial-notify) also mention
>>>
>>>> that a NOTIFY MAY be sent without a body.
>>>> Now, what a subscriber does when he receives a
>>>
>>>
>>> NOTIFY with
>>>
>>>> Content-Length: 0 in the middle of a subscription?
>>>
>>>
>>> Can he conclude
>>>
>>>> that the event state is not meaningful anymore and
>>>
>>>
>>> ditch the state
>>>
>>>> from previous NOTIFY? Or can he do that only if
>>>
>>>
>>> there is
>>>
>>>> a Content-Type header in the NOTIFY?
>>>>
>>>> --Pekka
>>>>
>>>> _______________________________________________
>>>> Simple mailing list
>>>> Simple@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/simple
>>>>
>>>
>>> --=20
>>> Jonathan D. Rosenberg, Ph.D.                600
>>> Lanidex Plaza
>>> Chief Technology Officer                   Parsippany, NJ 07054-2711
>>> dynamicsoft
>>> jdrosen@dynamicsoft.com                     FAX:  (973) 952-5050
>>> http://www.jdrosen.net                      PHONE:
>>> (973) 952-5000
>>> http://www.dynamicsoft.com
>>>
>>> _______________________________________________
>>> Simple mailing list
>>> Simple@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/simple
>>>
>>
>>
>>
>>
>>    =20
>>       =20
>> __________________________________
>> Do you Yahoo!?
>> New and Improved Yahoo! Mail - 100MB free storage!
>> http://promotions.yahoo.com/new_mail
>=20
>=20


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Whilst the design team is working on the security issues that were =
recently brought up on the list, we are asking the community to review =
the latest MSRP ID and send comments. The WGLC for MSRP will be issued =
as soon as the security concerns have been tackled and an new ID has =
been issued containing any modifications that may be required.

Bringing up any issues now will allow us to close as many open issues as =
soon as possible in parallel to solving the security ones. This will =
reduce the number of issues that will be brought up during the MSRP WGLC =
and will speed up the process of sending the draft to IESG (and =
hopefully keep the MSRP design team sane :)

The latest draft can be found at =
http://www.ietf.org/internet-drafts/draft-ietf-simple-message-sessions-08=
.txt

Regards,
Hisham

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From simple-bounces@ietf.org  Fri Sep 17 14:48:56 2004
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From: Eric Burger <eburger@brooktrout.com>
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Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE docu
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I can't agree with a statement that begins "a private extension...".

If the schema anticipates extension points (as Pete points out), then the
XML will be valid.

Saying "SHOULD be valid, because we might extend things willy-nilly without
standardization" MUST NOT be allowed.

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: Thursday, September 16, 2004 8:08 AM
> To: Jonathan Rosenberg; Pekka Pessi
> Cc: Robert Sparks; simple@ietf.org
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
> 
> 
> Please see below...
> ----- Original Message ----- 
> From: "Pekka Pessi" <Pekka.Pessi@nokia.com>
> 
> 
> > Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> > >Hmm, its a good question. I think that the vast majority of XML
> > >docs produced in the SIP/SIPPING/SIMPLE groups have the same
> > >words, and to be honest, I had not given a ton of thought to MUST
> > >vs. SHOULD for valid. I can't really think of a reason why it
> > >would ever be the right thing to produce a document that is not
> > >valid, particularly if the recipient does schema validation.
> >
> > The presence documents may contain all kind of extensions, which
> > may be only meaningful for a single application using the presence
> > service. The presence servers or any intermediate nodes (not to
> > speak about any innocent watcher) can't really know if the
> > documents are valid unless they know the schema for all of
> > those extensions.
> >
> > Like, if composer puts two tuples from two different sources to
> > the combined document in the presence server. Both the extensions
> > use ID attributes, but how could the composer know which
> > attributes have xs:ID type? The following example may be valid, or
> > then not, but who is to say without a schema?
> >
> > <?xml version="1.0" encoding="UTF-8"?>
> > <presence xmlns="urn:ietf:params:xml:ns:pidf"
> >      xmlns:x="http://example.com/ex/ten/sion"
> >      entity="pres:john.doe@example.com">
> >   <tuple id="id1"><status><basic>open</basic></status></tuple>
> >   <x:example-extension id="id1" />
> > </presence>
> >
> 
> 
> I don't know about the specifics of the presence schema, but 
> presumably the
> original presence schema specifies <xs:any 
> namespace='##other'/> at some key
> points.  Hence the document is valid against the original schema.
> (Mumbo-jumbo is valid in the schema sense as long as it is 
> from another
> namespace.)  Put another way, a document can be valid in the 
> schema sense,
> but still not be semantically correct.
> 
> How validity is established for the extensions is another 
> issue.  This comes
> down to versioning.  I think this is an area where the 
> group(s) needs to get
> a common understanding on how things should be done.  (Schema 
> is actually
> weak in this area!  There are ways, but they don't come for 
> free and they
> certainly don't happen by accident!)
> 
> Pete.
> --
> =============================================
> Pete Cordell
> Tech-Know-Ware
> pete(at)tech-know-ware.com
> http://www.tech-know-ware.com
> http://www.xml2cpp.com - Now Available
> +44 1473 635863
> =============================================
> 
> 
> > --Pekka
> >
> > >Robert Sparks wrote:
> ... cut ...
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

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Pekka Pessi wrote:

> Hi Jonathan,
> 
> 
>> If the Content-TYpe is present, and the length is zero, it means
>> that the body is of that type but has no information in it. I'll
>> note that this is probably illegal, since a PIDF document is
>> supposed to be valid/well-formed, and null content doesnt have that
>> property.
> 
> 
>> If the Content-Type is absent and the length is zero, it means that
>> there is no body, and the state of the watched resource is
>> unchanged.
> 
> 
> OK. So, if I understood the specs correctly, the notifier can send a
> NOTIFY without body at any time.

Right.

> 
> Now, let's assume that the subscriber is out of sync with, for 
> instance, partial presence notifications. How it can ask the server
> to send a NOTIFY with full state? It could terminate and recreate the
> subscription, but that potentially involve a large number of message.

A subscribe refresh is supposed to trigger a state update, for most 
packages.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Paul Kyzivat wrote:

> Jonathan,
> 
> What you suggest is plausible, but I don't recall ever seeing anything 
> written that discusses the meaning of a Content-Type with an empty body.
> 
> That logic would imply that there is (conceptually) a special content 
> type for which an empty body is valid and a non-empty body is not, and 
> that this is a default content-type when the body is empty.

I don't see how you would come to this conclusion. There is no default 
value for Content-Type.



> 
> I believe I have seen stacks that considered the presence of a content 
> type header invalid if there is no body. At the least there ought to be 
> something that says this is valid usage.

It is written, in RFC 3261, section 20.15:

20.15 Content-Type

    The Content-Type header field indicates the media type of the
    message-body sent to the recipient.  The "media-type" element is
    defined in [H3.7].  The Content-Type header field MUST be present if
    the body is not empty.  If the body is empty, and a Content-Type
    header field is present, it indicates that the body of the specific
    type has zero length (for example, an empty audio file).


-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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To: Jari Urpalainen <jari.urpalainen@nokia.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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inline.

Jari Urpalainen wrote:

> ext Jonathan Rosenberg wrote:
> 
>> inline.
>>
>> Pekka Pessi wrote:
>>
>>> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>>>
>>>> Hmm, its a good question. I think that the vast majority of XML
>>>> docs produced in the SIP/SIPPING/SIMPLE groups have the same
>>>> words, and to be honest, I had not given a ton of thought to MUST
>>>> vs. SHOULD for valid. I can't really think of a reason why it
>>>> would ever be the right thing to produce a document that is not
>>>> valid, particularly if the recipient does schema validation.
>>>
>>>
>>>
>>>
>>> The presence documents may contain all kind of extensions, which
>>> may be only meaningful for a single application using the presence
>>> service. The presence servers or any intermediate nodes (not to
>>> speak about any innocent watcher) can't really know if the
>>> documents are valid unless they know the schema for all of
>>> those extensions.
>>>
>>> Like, if composer puts two tuples from two different sources to
>>> the combined document in the presence server. Both the extensions
>>> use ID attributes, but how could the composer know which
>>> attributes have xs:ID type? The following example may be valid, or
>>> then not, but who is to say without a schema?
>>
>>
>>
>> It depends on how you want to define "valid". If a document has some 
>> extensions (say, rpid ones), such a document is still valid according 
>> to just the pidf schema, since it contains well-formed elements from 
>> another namespace in a place allowed by the schema.
>>
>> So, when we say "SHOULD be valid", do we mean that all elements in the 
>> document should be valid according to their defined schemas, or just 
>> that the document is valid according to the pidf schema?
>>
>> -Jonathan R.
> 
> 
> IMO pidf schema requirement is a "MUST" as content-type 
> application/pidf+xml is defined by that schema. How could you use some 
> instance document with that content-type if it is not valid ? 

No, that's not what I'm saying.

I'm saying that a document of type application/pidf+xml could be valid 
against the PIDF schema, but it contains extensions (which are allowed 
in the pidf schema), and those extensions make use of elements defined 
in another schema. However, those elements are not valid according to 
that other schema.

In this case, do we consider this document "valid"?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Mon, 20 Sep 2004 11:00:50 +0300
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	documents
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I see Jonathan's point here.=20

I think what we need is 2 normative sentences here: The XML document =
MUST be valid for the creator and SHOULD be valid for the consumer. It =
is a MUST for the creator since there is no reason whatsoever for the =
document to be produced and not validated. It is a SHOULD for the =
consumer since the XML document may have extensions that are not =
understood but are valid according to the base schema. The consumer can =
truly validate the XML document if he possesses all schema extensions =
required.

Regards,
Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Jonathan Rosenberg
> Sent: 20.September.2004 08:16
> To: Urpalainen Jari (Nokia-NRC/Helsinki)
> Cc: Pessi Pekka (Nokia-NRC/Helsinki); Robert Sparks; simple@ietf.org
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>=20
>=20
> inline.
>=20
> Jari Urpalainen wrote:
>=20
> > ext Jonathan Rosenberg wrote:
> >=20
> >> inline.
> >>
> >> Pekka Pessi wrote:
> >>
> >>> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> >>>
> >>>> Hmm, its a good question. I think that the vast majority of XML
> >>>> docs produced in the SIP/SIPPING/SIMPLE groups have the same
> >>>> words, and to be honest, I had not given a ton of thought to MUST
> >>>> vs. SHOULD for valid. I can't really think of a reason why it
> >>>> would ever be the right thing to produce a document that is not
> >>>> valid, particularly if the recipient does schema validation.
> >>>
> >>>
> >>>
> >>>
> >>> The presence documents may contain all kind of extensions, which
> >>> may be only meaningful for a single application using the presence
> >>> service. The presence servers or any intermediate nodes (not to
> >>> speak about any innocent watcher) can't really know if the
> >>> documents are valid unless they know the schema for all of
> >>> those extensions.
> >>>
> >>> Like, if composer puts two tuples from two different sources to
> >>> the combined document in the presence server. Both the extensions
> >>> use ID attributes, but how could the composer know which
> >>> attributes have xs:ID type? The following example may be valid, or
> >>> then not, but who is to say without a schema?
> >>
> >>
> >>
> >> It depends on how you want to define "valid". If a=20
> document has some=20
> >> extensions (say, rpid ones), such a document is still=20
> valid according=20
> >> to just the pidf schema, since it contains well-formed=20
> elements from=20
> >> another namespace in a place allowed by the schema.
> >>
> >> So, when we say "SHOULD be valid", do we mean that all=20
> elements in the=20
> >> document should be valid according to their defined=20
> schemas, or just=20
> >> that the document is valid according to the pidf schema?
> >>
> >> -Jonathan R.
> >=20
> >=20
> > IMO pidf schema requirement is a "MUST" as content-type=20
> > application/pidf+xml is defined by that schema. How could=20
> you use some=20
> > instance document with that content-type if it is not valid ?=20
>=20
> No, that's not what I'm saying.
>=20
> I'm saying that a document of type application/pidf+xml could=20
> be valid=20
> against the PIDF schema, but it contains extensions (which=20
> are allowed=20
> in the pidf schema), and those extensions make use of=20
> elements defined=20
> in another schema. However, those elements are not valid according to=20
> that other schema.
>=20
> In this case, do we consider this document "valid"?
>=20
> -Jonathan R.
>=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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From simple-bounces@ietf.org  Mon Sep 20 05:36:45 2004
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: <hisham.khartabil@nokia.com>, <jdrosen@dynamicsoft.com>,
        <Jari.Urpalainen@nokia.com>
References: <5816828233DEFA41807A6CFDFDF2343C08B613@esebe056.ntc.nokia.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Mon, 20 Sep 2004 10:27:29 +0100
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I think the statements needs to be something like:

The XML document MUST be valid for the creator (no change).

The XML document MUST be valid for the consumer according to the perhaps
incomplete set of schema copies that it holds (or has access to) that are
applicable to the XML document.

The wording sucks, but hopefully you can see the intent!

Pete.
--
=============================================
Pete Cordell
Tech-Know-Ware
pete(at)tech-know-ware.com
http://www.tech-know-ware.com
http://www.xml2cpp.com - Now Available
+44 1473 635863
=============================================

----- Original Message ----- 
From: <hisham.khartabil@nokia.com>
To: <jdrosen@dynamicsoft.com>; <Jari.Urpalainen@nokia.com>
Cc: <Pekka.Pessi@nokia.com>; <RjS@xten.com>; <simple@ietf.org>
Sent: Monday, September 20, 2004 9:00 AM
Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE
documents


I see Jonathan's point here.

I think what we need is 2 normative sentences here: The XML document MUST be
valid for the creator and SHOULD be valid for the consumer. It is a MUST for
the creator since there is no reason whatsoever for the document to be
produced and not validated. It is a SHOULD for the consumer since the XML
document may have extensions that are not understood but are valid according
to the base schema. The consumer can truly validate the XML document if he
possesses all schema extensions required.

Regards,
Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Jonathan Rosenberg
> Sent: 20.September.2004 08:16
> To: Urpalainen Jari (Nokia-NRC/Helsinki)
> Cc: Pessi Pekka (Nokia-NRC/Helsinki); Robert Sparks; simple@ietf.org
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>
>
> inline.
>
> Jari Urpalainen wrote:
>
> > ext Jonathan Rosenberg wrote:
> >
> >> inline.
> >>
> >> Pekka Pessi wrote:
> >>
> >>> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> >>>
> >>>> Hmm, its a good question. I think that the vast majority of XML
> >>>> docs produced in the SIP/SIPPING/SIMPLE groups have the same
> >>>> words, and to be honest, I had not given a ton of thought to MUST
> >>>> vs. SHOULD for valid. I can't really think of a reason why it
> >>>> would ever be the right thing to produce a document that is not
> >>>> valid, particularly if the recipient does schema validation.
> >>>
> >>>
> >>>
> >>>
> >>> The presence documents may contain all kind of extensions, which
> >>> may be only meaningful for a single application using the presence
> >>> service. The presence servers or any intermediate nodes (not to
> >>> speak about any innocent watcher) can't really know if the
> >>> documents are valid unless they know the schema for all of
> >>> those extensions.
> >>>
> >>> Like, if composer puts two tuples from two different sources to
> >>> the combined document in the presence server. Both the extensions
> >>> use ID attributes, but how could the composer know which
> >>> attributes have xs:ID type? The following example may be valid, or
> >>> then not, but who is to say without a schema?
> >>
> >>
> >>
> >> It depends on how you want to define "valid". If a
> document has some
> >> extensions (say, rpid ones), such a document is still
> valid according
> >> to just the pidf schema, since it contains well-formed
> elements from
> >> another namespace in a place allowed by the schema.
> >>
> >> So, when we say "SHOULD be valid", do we mean that all
> elements in the
> >> document should be valid according to their defined
> schemas, or just
> >> that the document is valid according to the pidf schema?
> >>
> >> -Jonathan R.
> >
> >
> > IMO pidf schema requirement is a "MUST" as content-type
> > application/pidf+xml is defined by that schema. How could
> you use some
> > instance document with that content-type if it is not valid ?
>
> No, that's not what I'm saying.
>
> I'm saying that a document of type application/pidf+xml could
> be valid
> against the PIDF schema, but it contains extensions (which
> are allowed
> in the pidf schema), and those extensions make use of
> elements defined
> in another schema. However, those elements are not valid according to
> that other schema.
>
> In this case, do we consider this document "valid"?
>
> -Jonathan R.
>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
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From simple-bounces@ietf.org  Mon Sep 20 10:36:48 2004
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Jonathan Rosenberg wrote:
> 
> 
> Paul Kyzivat wrote:
> 
>> Jonathan,
>>
>> What you suggest is plausible, but I don't recall ever seeing anything 
>> written that discusses the meaning of a Content-Type with an empty body.
>>
>> That logic would imply that there is (conceptually) a special content 
>> type for which an empty body is valid and a non-empty body is not, and 
>> that this is a default content-type when the body is empty.
> 
> I don't see how you would come to this conclusion. There is no default 
> value for Content-Type.

This is probably just splitting hairs, and not worth discussion.

>> I believe I have seen stacks that considered the presence of a content 
>> type header invalid if there is no body. At the least there ought to 
>> be something that says this is valid usage.
> 
> 
> It is written, in RFC 3261, section 20.15:
> 
> 20.15 Content-Type
> 
>    The Content-Type header field indicates the media type of the
>    message-body sent to the recipient.  The "media-type" element is
>    defined in [H3.7].  The Content-Type header field MUST be present if
>    the body is not empty.  If the body is empty, and a Content-Type
>    header field is present, it indicates that the body of the specific
>    type has zero length (for example, an empty audio file).

OK - I didn't remember that was there.

	Paul


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hisham.khartabil@nokia.com wrote:
> I see Jonathan's point here. 
> 
> I think what we need is 2 normative sentences here:
 > The XML document MUST be valid for the creator

I think the above can cause problems, depending on how you define "creator".

Consider a presence document that is created by a UA and published to a 
PS. The PS then aggregates the contents of this published document, 
filters that, and thus creates another presence document that is send 
out in a NOTIFY.

The UA could have used an extension that is unknown to the PS. But IMO 
the PS should still have the option of including that unknown portion 
into the document it sends out in the NOTIFY. But then it would be 
violating this new proposed rule.

Imposing this rule *requires* that the PS be aware of any extensions 
used by any of its sources of presence information.

So I think this all needs to be SHOULD level. Also, I agree with Pete's 
later comment that validity according to PIDF is always a requirement.

	Paul

 > and SHOULD be valid for the consumer. It is a MUST for the creator 
since there is no reason whatsoever for the document to be produced and 
not validated. It is a SHOULD for the consumer since the XML document 
may have extensions that are not understood but are valid according to 
the base schema. The consumer can truly validate the XML document if he 
possesses all schema extensions required.
> 
> Regards,
> Hisham
> 
> 
>>-----Original Message-----
>>From: simple-bounces@ietf.org 
>>[mailto:simple-bounces@ietf.org]On Behalf
>>Of ext Jonathan Rosenberg
>>Sent: 20.September.2004 08:16
>>To: Urpalainen Jari (Nokia-NRC/Helsinki)
>>Cc: Pessi Pekka (Nokia-NRC/Helsinki); Robert Sparks; simple@ietf.org
>>Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
>>documents
>>
>>
>>inline.
>>
>>Jari Urpalainen wrote:
>>
>>
>>>ext Jonathan Rosenberg wrote:
>>>
>>>
>>>>inline.
>>>>
>>>>Pekka Pessi wrote:
>>>>
>>>>
>>>>>Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>>>>>
>>>>>
>>>>>>Hmm, its a good question. I think that the vast majority of XML
>>>>>>docs produced in the SIP/SIPPING/SIMPLE groups have the same
>>>>>>words, and to be honest, I had not given a ton of thought to MUST
>>>>>>vs. SHOULD for valid. I can't really think of a reason why it
>>>>>>would ever be the right thing to produce a document that is not
>>>>>>valid, particularly if the recipient does schema validation.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>The presence documents may contain all kind of extensions, which
>>>>>may be only meaningful for a single application using the presence
>>>>>service. The presence servers or any intermediate nodes (not to
>>>>>speak about any innocent watcher) can't really know if the
>>>>>documents are valid unless they know the schema for all of
>>>>>those extensions.
>>>>>
>>>>>Like, if composer puts two tuples from two different sources to
>>>>>the combined document in the presence server. Both the extensions
>>>>>use ID attributes, but how could the composer know which
>>>>>attributes have xs:ID type? The following example may be valid, or
>>>>>then not, but who is to say without a schema?
>>>>
>>>>
>>>>
>>>>It depends on how you want to define "valid". If a 
>>>
>>document has some 
>>
>>>>extensions (say, rpid ones), such a document is still 
>>>
>>valid according 
>>
>>>>to just the pidf schema, since it contains well-formed 
>>>
>>elements from 
>>
>>>>another namespace in a place allowed by the schema.
>>>>
>>>>So, when we say "SHOULD be valid", do we mean that all 
>>>
>>elements in the 
>>
>>>>document should be valid according to their defined 
>>>
>>schemas, or just 
>>
>>>>that the document is valid according to the pidf schema?
>>>>
>>>>-Jonathan R.
>>>
>>>
>>>IMO pidf schema requirement is a "MUST" as content-type 
>>>application/pidf+xml is defined by that schema. How could 
>>
>>you use some 
>>
>>>instance document with that content-type if it is not valid ? 
>>
>>No, that's not what I'm saying.
>>
>>I'm saying that a document of type application/pidf+xml could 
>>be valid 
>>against the PIDF schema, but it contains extensions (which 
>>are allowed 
>>in the pidf schema), and those extensions make use of 
>>elements defined 
>>in another schema. However, those elements are not valid according to 
>>that other schema.
>>
>>In this case, do we consider this document "valid"?
>>
>>-Jonathan R.
>>
>>
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>>
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 


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From simple-bounces@ietf.org  Mon Sep 20 12:53:54 2004
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inline.

Paul Kyzivat wrote:

> 
> 
> hisham.khartabil@nokia.com wrote:
> 
>> I see Jonathan's point here.
>> I think what we need is 2 normative sentences here:
> 
>  > The XML document MUST be valid for the creator
> 
> I think the above can cause problems, depending on how you define 
> "creator".
> 
> Consider a presence document that is created by a UA and published to a 
> PS. The PS then aggregates the contents of this published document, 
> filters that, and thus creates another presence document that is send 
> out in a NOTIFY.
> 
> The UA could have used an extension that is unknown to the PS. But IMO 
> the PS should still have the option of including that unknown portion 
> into the document it sends out in the NOTIFY. But then it would be 
> violating this new proposed rule.

Right. It can be easier to define these validity rules in the context of 
a specific protocol usage, and in the above case we could except a 
presence server from needing to generate valid documents. Those kinds of 
rules belong in protocol documents, not in the specifications of the 
document formats themselves.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: [Simple] FW: CPCP Issue 37: Generic privileges solution
Date: Tue, 21 Sep 2004 10:57:56 +0300
Message-ID: <5816828233DEFA41807A6CFDFDF2343C08B621@esebe056.ntc.nokia.com>
Thread-Topic: CPCP Issue 37: Generic privileges solution
Thread-Index: AcSZl/3YES5bsnqwSe+ryN/uBrPDqgAAL7cAAYX55dA=
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It would be great if I can get people's gut feeling on this.

Thanks,
Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext hisham.khartabil@nokia.com
> Sent: 13.September.2004 16:52
> To: simple@ietf.org
> Subject: [Simple] FW: CPCP Issue 37: Generic privileges solution
>=20
>=20
> The draft can be found at:
> http://www.imsbook.com/hisham/draft-ietf-xcon-conference-polic
> y-privileges-00.txt
>=20
> SIMPLE WG comments are being sorted on this issue presented below.
>=20
> Thanks,
> Hisham
>=20
> > -----Original Message-----
> > From: Khartabil Hisham (Nokia-TP-MSW/Helsinki)=20
> > Sent: 13.September.2004 16:46
> > To: 'xcon@ietf.org'
> > Subject: CPCP Issue 37: Generic privileges solution
> >=20
> >=20
> > This issue affects=20
> > draft-ietf-xcon-conference-policy-privileges-00.txt.
> >=20
> > The schema in this draft is generic and can be used for any=20
> > XML document that carries user information and resides on a=20
> > server. It is useful especially for XCAP.
> >=20
> > I have been toying with the idea of extracting the schema=20
> > into a separate draft and submitting that in SIMPLE. XCON can=20
> > then use that draft and customise it for conferencing needs.=20
> > By customising it, I mean defining the actions and=20
> > transformations that grant the conferencing specific=20
> > privileges to users.
> >=20
> > This solution uses common policy as well where the conditions=20
> > carry the document URI and the identity of users and the=20
> > actions carry the privileges (read/modify certain parts of=20
> > the conference policy).
> >=20
> > BTW, the draft defines a sematic approach as agreed in San Diego.
> >=20
> > So, what do people think of that?
> >=20
> > I'll forward this email to SIMPLE for comments.
> >=20
> > Regards,
> > Hisham
> >=20
>=20
> _______________________________________________
> Simple mailing list
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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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Subject: [Simple] MSRP URLs
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Hi:

A few comments about section 5 of the MSRP draft:

http://www.ietf.org/internet-drafts/draft-ietf-simple-message-sessions-08.txt

1) The draft should describe the type of resources that are identified 
by an MSRP URL. So I would suggest to add an introductory paragraph at 
the beginning of section 5 that says something on the following lines:

    The MSRP and MSRPS URL schemes are used to designate a session of
    instant messages at a particular host computer.

   Question: is this definition also applicable to relays?

2) RFC 2396 is under revision, nowadays 
draft-fielding-uri-rfc2396bis-06.txt. I am not sure if you still want to 
refer to RFC 2396 or the revision of it. I guess this depends of which 
draft makes it first to the IESG.

Among other things, in case you want to refer to rfc2396bis, this draft 
has removed the definition of "hostport". Now it is just "host [: port]"

3) The current definition of an MSRP URL includes a "resource" part, 
when actually it is a "path" or "session-id", because a "resource" is 
the the whole MSRP URL, including the URI scheme and the hostname.

So I would suggest to replace the current definition:

     MSRP_urls = msrp-scheme "://" [userinfo "@"] hostport ["/"
       resource] ";" transport
       resource = 1*unreserved

with this other one:

     MSRP_urls = msrp-scheme "://" [userinfo "@"] hostport ["/"
       path] ";" transport
       resource = 1*unreserved

4) According to rfc2396bis, "unrerved" is defined as:

unreserved    = ALPHA / DIGIT / "-" / "." / "_" / "~"

On purpose "unreserved" does not include "sub-delims", which is defined as:

     sub-delims    = "!" / "$" / "&" / "'" / "(" / ")"
                   / "*" / "+" / "," / ";" / "="

I guess it is OK for MSRP that the path is formed of unreserved 
characters, not including sub-delims. But I would like to point this out 
in case someone feels the path needs to include sub-delims.

5) I find a contradiction that we, on one hand, allow to have a 
"userinfo" part in the MSRP URL including a reference to RFC 2396, and 
on the other hand we restrict the "userinfo" definition in RFC 23964 to 
unreserved characters.

RFC 2396 (and 2396bis) allow to have escaped characters in the userinfo. 
However, MSRP does not, but still refers to 2396 for the definition of 
userinfo.

So, if the "userinfo" defined in the MSRP URL is different from (or a 
subset of) the "userinfo" defined in RFC 2396, then I would suggest to 
change the name (perhaps "unescaped-userinfo"), and avoid the reference 
to RFC 2396.

6) Section 5.1 reads on the second bullet point:

    2.  If the hostpart contains an eplicit IP address, and/or port,
        these are compared numerically.  Otherwise, hostpart is compared
        as a case insensitive character string.

I was wondering in the how to treat zeros in either IPv4 or IPv6 (IPv6 
uses this quite oftenly). The missleading word is "numerically", since I 
am not sure if the "::" should be normalized by adding trailing and 
leading zeros or not.

Is it the intention that the following two URLs are equivalent and MSRP 
endpoints should detect it?

    msrp://[1080:0:0:0:8:800:200C:417A]/aodisfa;tcp
    msrp://[1080::8:800:200C:417A]/aodisfa;tcp

A clarification should be written.


Regards,

         Miguel


-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland

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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
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	message of "Mon, 20 Sep 2004 01:16:09 -0400")
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Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
>I'm saying that a document of type application/pidf+xml could be valid
>against the PIDF schema, but it contains extensions (which are allowed in
>the pidf schema), and those extensions make use of elements defined in
>another schema. However, those elements are not valid according to that
>other schema.

>In this case, do we consider this document "valid"?

Term valid has at least three different senses with the XML:

1) 

W3C defines that an XML document is valid when it has a DTD and
the document complies with the constraints set forth in DTD.

That is not what we want. No <!DOCTYPE> for presence, no "MUST be
valid", please.

(When RFC 3863 says that the application/pidf+xml document MUST
be well-formed and SHOULD be valid, I guess it just says that
extensions are allowed, even if they are not shown in the DTD.)

2) 

XML schema specification uses a bit different terminology. Schemas
don't specify documents, but trees (item and all its descendants). 
You can assess validity of a particular tree, for instance, all
the XML stuff within <presence> element. The validator can also
skip certain elements (like the wildcard elements used for
extending pidf). So, validator can return different results: tree
and items within it are fully validated, partially validated, or
not validated at all. Also each, item within the tree are valid or
invalid or their validity is not known.  

I'd say that the XML document is schema-valid if its root element
is fully validated and it and all its descendants are valid. This
is not the official definition, however.

This is unfeasible for the intermediate nodes (or processes), too. 
Unless the node knows the schema (and semantics) of all the
extensions, it cannot assess the validity of composed or filtered
documents. I think filtering and composing is precisely the things
that we want to do in the server nodes.

I can also imagine some applications that would just use existing
presence service, just add their non-standard application-specific
status in the published presence document. Now imagine two of
those applications running on the same node.

3)

Beside validity assessment, XML schema specification has the
concept of "local schema-validity", too:

<http://www.w3.org/TR/xmlschema-1/#section-Overview-of-XML-Schema>

I must admit that I don't fully understand what this
means, but I have a good hunch that it is what most of us
want. ;)

A element or attribute is valid if it satisfies the constraints
set by the XML schema. So, <presence> element is valid, if it
contains entity attribute, zero or more <tuple> elements, then
zero or more <note> elements and after then something wildcarded
stuff (I guess that is anything else but pidf-namespace stuff). I
don't understand how the datatype checking of evil types like
xs:ID or xs:IDREF is done when the "local schema-validity" is
determined. I *guess* that ID/IDREF attribute in tuple is valid if
its value is syntactically correct. Like id="_123" is valid but,
id="123" is not. No check for all id's within a document/item and
all of its descendats is required.

So, if we have a application/pidf+xml document, we can simply say
that it MUST conform to its schema and other requirements set
forth in RFC 3863. If we want to be more verbose, we could say
that the root node in the document MUST be partially or fully
validated and none of the validated nodes MUST NOT be invalid. Or
we could invent our own term and say that documents MUST be
locally-schema-valid with the schema XXX.

--Pekka

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Subject: [Simple] Scope of XCAP PIDF manipulation wrt. Presence Data Model
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Hi all,

The approach presented in Data Model for Presence =
(http://www.ietf.org/internet-drafts/draft-rosenberg-simple-presence-data=
-model-00.txt) has been in principle approved in SIMPLE WG, but there =
are still some open issues on details. One of them concerns the scope of =
XCAP application usage for PIDF manipulation =
(http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-pidf-manipula=
tion-usage-01.txt). The Data Model draft has the following text in =
Chapter 10:=20

      One of the source of confusion around the XCAP manipulation of
      PIDF [17] is that it was unclear as to why one would use it as
      opposed to PUBLISH.  The presence data model presented here sheds
      some light on that.  PUBLISH is appropriate for communicating
      information about services and devices.  PUBLISH is designed for
      the model where there can be more than one such source (as there
      is for devices and servcies), and where such state is soft-state
      (as it should be for devices and services).  However, presentity
      state is not clearly soft-state, and the model here clearly
      indicates that each presence document can have a single presentity
      element.  Thus, it might make sense to change the
      pidf-manipulation spec to only allow setting of presentity tuples.
      Now, that doesnt forbid a source from trying to PUBLISH presentity
      information, but there is clearly a need for a hard-state approach
      for setting presentity information.

The concrete issue here is whether XCAP PIDF manipulation should be =
restricted to setting only presentity tuple, i.e. not allow the setting =
of service tuples or device information.

My opinion is that we should not make this kind of restriction.=20

There is at least one use case where "hard state" service tuples make a =
lot of sense, and that is service tuples for "persistent" services such =
as e-mail, SMS, MMS or voicemail. These communication means are =
available for the sender even if the recipient is not on-line wrt. that =
service, since there is a network-based "inbox" taking care of them. =
XCAP PIDF manipulation is a better way to describe this kind "off-line" =
state for such services rather than SIP PUBLISH. (BTW, I believe we =
should define some new status attribute to this kind of services to =
clearly indicate the difference between "MMS available but no device =
on-line" vs. "MMS available with a device on-line" etc. Or probably the =
standardization fora responsible for e.g. MMS can do that.)

What might be a useful specification is to say that in a typical =
scenario information derived from SIP PUBLISH takes presedence over the =
information derived from XCAP. But that is already in the territory of =
specifying the composer policy, which I think is a separate effort from =
the data model of XCAP PIDF manipulation drafts.

So my proposal is basically to keep the XCAP PIDF manipulation draft as =
it is at the moment, and not make any restricting statements about its =
use in the Data Model either. Are there other opinions on this topic?

Regards,
	Markus
=20

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Hi Jonathan, Paul,

According to the following, rfc 2045 defines a default content type of  
text/plain; charset=us-ascii

Cheers,
Alex.

===========================================================================
RFC 2045                Internet Message Bodies            November 1996


5.2.  Content-Type Defaults

   Default RFC 822 messages without a MIME Content-Type header are taken
   by this protocol to be plain text in the US-ASCII character set,
   which can be explicitly specified as:

     Content-type: text/plain; charset=us-ascii

   This default is assumed if no Content-Type header field is specified.
   It is also recommend that this default be assumed when a
   syntactically invalid Content-Type header field is encountered. In
   the presence of a MIME-Version header field and the absence of any
   Content-Type header field, a receiving User Agent can also assume
   that plain US-ASCII text was the sender's intent.  Plain US-ASCII
   text may still be assumed in the absence of a MIME-Version or the
   presence of an syntactically invalid Content-Type header field, but
   the sender's intent might have been otherwise.

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

Paul Kyzivat wrote:

>
>
> Jonathan Rosenberg wrote:
>
>>
>>
>> Paul Kyzivat wrote:
>>
>>> Jonathan,
>>>
>>> What you suggest is plausible, but I don't recall ever seeing 
>>> anything written that discusses the meaning of a Content-Type with 
>>> an empty body.
>>>
>>> That logic would imply that there is (conceptually) a special 
>>> content type for which an empty body is valid and a non-empty body 
>>> is not, and that this is a default content-type when the body is empty.
>>
>>
>> I don't see how you would come to this conclusion. There is no 
>> default value for Content-Type.
>
>
> This is probably just splitting hairs, and not worth discussion.
>
>>> I believe I have seen stacks that considered the presence of a 
>>> content type header invalid if there is no body. At the least there 
>>> ought to be something that says this is valid usage.
>>
>>
>>
>> It is written, in RFC 3261, section 20.15:
>>
>> 20.15 Content-Type
>>
>>    The Content-Type header field indicates the media type of the
>>    message-body sent to the recipient.  The "media-type" element is
>>    defined in [H3.7].  The Content-Type header field MUST be present if
>>    the body is not empty.  If the body is empty, and a Content-Type
>>    header field is present, it indicates that the body of the specific
>>    type has zero length (for example, an empty audio file).
>
>
> OK - I didn't remember that was there.
>
>     Paul
>
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple



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Alex Audu wrote:
> Hi Jonathan, Paul,
> 
> According to the following, rfc 2045 defines a default content type of  
> text/plain; charset=us-ascii

Well, if we assume the same applies to SIP (not necessarily the case, 
but plausible), then in the absence of any Content- headers:

- the Content-Type would default to 'text/plain'
- the Content-Disposition would default to 'render'

So it will amount to a request to render a null string of text.

If this is applied to NOTIFY, it *could* result in a 415 response.

Seems like the right thing to do would be for each event type where this 
can happen to define the null body that is meaningful. Either explicitly 
note a null body of type text/plain, or else define the relevant mime 
type to include a null body.

	Paul

> Cheers,
> Alex.
> 
> ===========================================================================
> RFC 2045                Internet Message Bodies            November 1996
> 
> 
> 5.2.  Content-Type Defaults
> 
>   Default RFC 822 messages without a MIME Content-Type header are taken
>   by this protocol to be plain text in the US-ASCII character set,
>   which can be explicitly specified as:
> 
>     Content-type: text/plain; charset=us-ascii
> 
>   This default is assumed if no Content-Type header field is specified.
>   It is also recommend that this default be assumed when a
>   syntactically invalid Content-Type header field is encountered. In
>   the presence of a MIME-Version header field and the absence of any
>   Content-Type header field, a receiving User Agent can also assume
>   that plain US-ASCII text was the sender's intent.  Plain US-ASCII
>   text may still be assumed in the absence of a MIME-Version or the
>   presence of an syntactically invalid Content-Type header field, but
>   the sender's intent might have been otherwise.
> 
> ====================================================================
> 
> Paul Kyzivat wrote:
> 
>>
>>
>> Jonathan Rosenberg wrote:
>>
>>>
>>>
>>> Paul Kyzivat wrote:
>>>
>>>> Jonathan,
>>>>
>>>> What you suggest is plausible, but I don't recall ever seeing 
>>>> anything written that discusses the meaning of a Content-Type with 
>>>> an empty body.
>>>>
>>>> That logic would imply that there is (conceptually) a special 
>>>> content type for which an empty body is valid and a non-empty body 
>>>> is not, and that this is a default content-type when the body is empty.
>>>
>>>
>>>
>>> I don't see how you would come to this conclusion. There is no 
>>> default value for Content-Type.
>>
>>
>>
>> This is probably just splitting hairs, and not worth discussion.
>>
>>>> I believe I have seen stacks that considered the presence of a 
>>>> content type header invalid if there is no body. At the least there 
>>>> ought to be something that says this is valid usage.
>>>
>>>
>>>
>>>
>>> It is written, in RFC 3261, section 20.15:
>>>
>>> 20.15 Content-Type
>>>
>>>    The Content-Type header field indicates the media type of the
>>>    message-body sent to the recipient.  The "media-type" element is
>>>    defined in [H3.7].  The Content-Type header field MUST be present if
>>>    the body is not empty.  If the body is empty, and a Content-Type
>>>    header field is present, it indicates that the body of the specific
>>>    type has zero length (for example, an empty audio file).
>>
>>
>>
>> OK - I didn't remember that was there.
>>
>>     Paul
>>
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
> 
> 
> 
> 


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Date: Tue, 21 Sep 2004 11:41:50 -0700
Subject: Re: [Simple] MSRP Security Review: RFC3862 Application Considerations
From: Cullen Jennings <fluffy@cisco.com>
To: <hisham.khartabil@nokia.com>, Paul H Kyzivat <pkyzivat@cisco.com>
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Seem to me that multiple To, Subject, Foo etc in CPIM is not an issues for
MSRP - MSRP caries CPIM - headers in CPIM mean what CPIM says they mean. My
vague read of 3862 is that there can me one From , multiple To, multiple cc,
one DateTime, and is though it is unclear on Subject, I would read it as
only one Subject since the things that are multiple it explicitly mentions
as such. 


On 9/10/04 8:55 AM, "hisham.khartabil@nokia.com"
<hisham.khartabil@nokia.com> wrote:

> 
> 
>> -----Original Message-----
>> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: 10.September.2004 16:26
>> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
>> Cc: ben@nostrum.com; simple@ietf.org
>> Subject: Re: [Simple] MSRP Security Review: RFC3862 Application
>> Considerations
>> 
>> 
>> Hisham,
>> 
>> I agree with most of your comments, but I question the logic
>> of multiple 
>> Subject headers. What does this mean? Is the actual subject the
>> catenation of the contents of all the Subject headers? Should
>> they have 
>> <CR><LF> between then when displayed? Or are they to be treated as
>> alternatives, such as the same subject represented in
>> different languages?
> 
> That's what I had in mind.
> 
> /Hisham
> 
>> 
>> I believe 3261 only allows one Subject header. I don't know
>> about email, 
>> but by common usage I don't think it would work well if there
>> was more 
>> than one.
>> 
>> Paul
>> 
>> hisham.khartabil@nokia.com wrote:
>>> 
>>>> -----Original Message-----
>>>> From: simple-bounces@ietf.org
>>>> [mailto:simple-bounces@ietf.org]On Behalf
>>>> Of ext Ben Campbell
>>>> Sent: 10.September.2004 05:22
>>>> To: Simple WG
>>>> Subject: [Simple] MSRP Security Review: RFC3862 Application
>>>> Considerations
>>>> 
>>>> 
>>>> Hi,
>>>> 
>>>> Sam's security review pointed out that we had not handled the
>>>> application considerations required by RFC3862 on all applications
>>>> using the message/cpim MIME type. Here is a first cut at those
>>>> considerations for discussion (quoted text from RFC3862)
>>>> 
>>>> 
>>>>>  Applications using this specification must also specify:
>>>>> 
>>>>>   o  all headers that must be recognized by implementations of the
>>>>>      application
>>>> 
>>>> All MSRP endpoints MUST recognize the From, To, DateTime,
>> and Require 
>>>> headers as defined in RFC3862. Such applications SHOULD
>> recognize the 
>>>> CC header, and MAY recognize the Subject header. Any MSRP
>> application 
>>>> that recognizes any message/cpim header MUST understand the
>> NS (name 
>>>> space) header.
>>>> 
>>>> 
>>>>>   o  any headers that must be present in all messages
>>>> 
>>>> created by that
>>>> 
>>>>>      application.
>>>> 
>>>> All message/cpim body parts send by an MSRP endpoint MUST
>> include the 
>>>> From and To headers. If the message/cpim body part is
>>>> protected using
>>>> S/MIME, then it SHOULD also include the DateTime header.
>>>> 
>>>> 
>>>>>   o  any headers that may appear more than once in a
>>>> 
>>>> message, and how
>>>> 
>>>>>      they are to be interpreted (e.g., how to interpret multiple
>>>>>      'Subject:' headers with different language parameter values).
>>>>> 
>>>> 
>>>> Message/cpim headers defined in RFC 3862, with the exception
>>>> of the NS 
>>>> header, MUST NOT occur more than once in a given message/cpim
>>>> body part 
>>>> in an MSRP message. The Require header MAY include multiple
>>>> values. The 
>>>> NS header MAY occur zero or more times, depending on how many name
>>>> spaces are being referenced.
>>>> 
>>>> Extension headers MAY occur more than once, depending on the
>>>> definition 
>>>> of such headers.
>>> 
>>> 
>>> The To-header field should not be limited to 1, nor should
>> the CC field. This could be used in the future for, what is
>> commonly referred to as, whispering or private messages
>> within a conference.
>>> 
>>> Also, the subject header might appear multiple times. I
>> would say we limit the from-header to one and no limit on the rest.
>>> 
>>> Regards,
>>> Hisham
>>> 
>>> 
>>>>>   o  Security mechanisms and crytography schemes to be
>>>> 
>>>> used with the
>>>> 
>>>>>      application, including any mandatory-to-implement security
>>>>>      provisions.
>>>> 
>>>> [Ben: I _think_ we are covering this in general--do we need to
>>>> explicitly relate such mechanisms to message/cpim?]
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Simple mailing list
>>>> Simple@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/simple
>>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Simple mailing list
>>> Simple@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/simple
>>> 
>> 
>> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple



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Subject: RE: [Simple] NOTIFY without message-body
Date: Wed, 22 Sep 2004 10:47:11 +0300
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> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Jonathan Rosenberg
> Sent: 20.September.2004 08:04
> To: Paul Kyzivat
> Cc: Pessi Pekka (Nokia-NRC/Helsinki); simple@ietf.org
> Subject: Re: [Simple] NOTIFY without message-body
>=20
>=20
>=20
>=20
> Paul Kyzivat wrote:
>=20
> > Jonathan,
> >=20
> > What you suggest is plausible, but I don't recall ever=20
> seeing anything=20
> > written that discusses the meaning of a Content-Type with=20
> an empty body.
> >=20
> > That logic would imply that there is (conceptually) a=20
> special content=20
> > type for which an empty body is valid and a non-empty body=20
> is not, and=20
> > that this is a default content-type when the body is empty.
>=20
> I don't see how you would come to this conclusion. There is=20
> no default=20
> value for Content-Type.

I agree here. Also, the presence of Content-* header does not mean there =
is a body. Take for example Content-Length. Many implementations set =
that to 0. This could be 2 things:

1. There is no body
2. There is a body, but it has a length of 0.

If the interperetation is 2, then content-type needs to have default. We =
have so have assumed that content-length: 0 means no body (ie. 1 above).

>=20
>=20
>=20
> >=20
> > I believe I have seen stacks that considered the presence=20
> of a content=20
> > type header invalid if there is no body. At the least there=20
> ought to be=20
> > something that says this is valid usage.
>=20
> It is written, in RFC 3261, section 20.15:
>=20
> 20.15 Content-Type
>=20
>     The Content-Type header field indicates the media type of the
>     message-body sent to the recipient.  The "media-type" element is
>     defined in [H3.7].  The Content-Type header field MUST be=20
> present if
>     the body is not empty.  If the body is empty, and a Content-Type
>     header field is present, it indicates that the body of=20
> the specific
>     type has zero length (for example, an empty audio file).

This may result in a 400 response being returned for, say, XML bodies =
since they are not valid (actually, it is unclear what the exact SIP =
error response for a malformed body should be. This was an issue at the =
last SIPit).

Regards,
Hisham

>=20
>=20
> -Jonathan R.
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Wed, 22 Sep 2004 11:54:06 +0300
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How does this proposal sound (for the filter drafts):

In the format document, the following text appears: The XML document =
MUST be well-formed and MUST be valid according to schemas, including =
extensions schemas, available to the validater and applicable to the XML =
document.

In the functionality draft: The creator MUST ensure that the XML =
document is well-formed and valid. The creator MUST NOT insert any =
extension elements or attributes into the XML document unless it has =
access to the extension schema and can validate the XML document. The =
XML document consumer MUST validate the XML document according the =
schemas, including extension schemas, it has access to that are =
applicable to this XML document.

For pidf, the second part might be slightly different since the creator =
of the document can be the compositor and the compositor can create a =
presence XML document by aggregating tuples that contain extensions that =
the compositor is unaware of. Perhaps something can go in the Date Model =
draft discussing this.

Regards,
Hisham



> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Pekka Pessi
> Sent: 20.September.2004 14:36
> To: Jonathan Rosenberg
> Cc: Pessi Pekka (Nokia-NRC/Helsinki); Urpalainen Jari
> (Nokia-NRC/Helsinki); simple@ietf.org; Robert Sparks
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>=20
>=20
> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> >I'm saying that a document of type application/pidf+xml=20
> could be valid
> >against the PIDF schema, but it contains extensions (which=20
> are allowed in
> >the pidf schema), and those extensions make use of elements=20
> defined in
> >another schema. However, those elements are not valid=20
> according to that
> >other schema.
>=20
> >In this case, do we consider this document "valid"?
>=20
> Term valid has at least three different senses with the XML:
>=20
> 1)=20
>=20
> W3C defines that an XML document is valid when it has a DTD and
> the document complies with the constraints set forth in DTD.
>=20
> That is not what we want. No <!DOCTYPE> for presence, no "MUST be
> valid", please.
>=20
> (When RFC 3863 says that the application/pidf+xml document MUST
> be well-formed and SHOULD be valid, I guess it just says that
> extensions are allowed, even if they are not shown in the DTD.)
>=20
> 2)=20
>=20
> XML schema specification uses a bit different terminology. Schemas
> don't specify documents, but trees (item and all its descendants).=20
> You can assess validity of a particular tree, for instance, all
> the XML stuff within <presence> element. The validator can also
> skip certain elements (like the wildcard elements used for
> extending pidf). So, validator can return different results: tree
> and items within it are fully validated, partially validated, or
> not validated at all. Also each, item within the tree are valid or
> invalid or their validity is not known. =20
>=20
> I'd say that the XML document is schema-valid if its root element
> is fully validated and it and all its descendants are valid. This
> is not the official definition, however.
>=20
> This is unfeasible for the intermediate nodes (or processes), too.=20
> Unless the node knows the schema (and semantics) of all the
> extensions, it cannot assess the validity of composed or filtered
> documents. I think filtering and composing is precisely the things
> that we want to do in the server nodes.
>=20
> I can also imagine some applications that would just use existing
> presence service, just add their non-standard application-specific
> status in the published presence document. Now imagine two of
> those applications running on the same node.
>=20
> 3)
>=20
> Beside validity assessment, XML schema specification has the
> concept of "local schema-validity", too:
>=20
> <http://www.w3.org/TR/xmlschema-1/#section-Overview-of-XML-Schema>
>=20
> I must admit that I don't fully understand what this
> means, but I have a good hunch that it is what most of us
> want. ;)
>=20
> A element or attribute is valid if it satisfies the constraints
> set by the XML schema. So, <presence> element is valid, if it
> contains entity attribute, zero or more <tuple> elements, then
> zero or more <note> elements and after then something wildcarded
> stuff (I guess that is anything else but pidf-namespace stuff). I
> don't understand how the datatype checking of evil types like
> xs:ID or xs:IDREF is done when the "local schema-validity" is
> determined. I *guess* that ID/IDREF attribute in tuple is valid if
> its value is syntactically correct. Like id=3D"_123" is valid but,
> id=3D"123" is not. No check for all id's within a document/item and
> all of its descendats is required.
>=20
> So, if we have a application/pidf+xml document, we can simply say
> that it MUST conform to its schema and other requirements set
> forth in RFC 3863. If we want to be more verbose, we could say
> that the root node in the document MUST be partially or fully
> validated and none of the validated nodes MUST NOT be invalid. Or
> we could invent our own term and say that documents MUST be
> locally-schema-valid with the schema XXX.
>=20
> --Pekka
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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Subject: [Simple] Presence Data Model: Identifying services
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Hi all,

I have a comment on Presence Data Model draft about the =
identification/characterization of services. (I know this has been =
discussed for a while already, but I still think the current way is =
somewhat unadequate.)

The current Data Model draft says:

   In this model, services are not explicitly enumerated.  That is,
   there is no "service" attribute with values of "ptt" or "telephony".
   Rather, the service is identified in one of two ways.  In many cases,
   the URI scheme is a clear indicator of service.  An "sms" URI clearly
   indicates SMS, and a "tel" URI clearly indicates telephony.  For some
   URIs, there may be many services available, for example, SIP.  For
   those services, each service has a set of characteristics, each of
   which has a well-defined meaning, such that a system can
   unequivocally determine whether or not the service has that
   characteristic.  This is discussed in more detail in [5].=20

I agree that the contact URI scheme is the most important piece of =
information to distinguish what is the "service". (There are some gaps =
here too, since e.g. the URI schemes for SMS or MMS do not even exist, =
and there may be other non-IETF protocols that face the same problem.) =
However, I'm not convinced that listing the signaling/media =
characteristics of the end-point or service really gives enough =
information to the watcher to really determine if it can communicate =
with the advertised service/application.=20

The refenced draft [5] has a following example:

5.6 Walkie-talkie

   The walkie-talkie service allows real-time voice communication
   between participants.  Only one participant can speak at a time; that
   is, communication is half-duplex.  Typically, participants press a
   button to indicate that they are ready to speak, although other
   mechanism (e.g.  voice activation) are occasionally used.

   Support for the Walkie-Talkie service can be deduced by observing the
   presence of a contact URI with a scheme of "sip:", associated with
   the following minimal set of capabilities: <audio>true</audio>
   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>

Presumably this is the same as the service sometimes called Push-to-Talk =
(PTT, PoC). The concrete problem is that there are several =
_non-interworking_ variants of this service, that still all would match =
the characteristics listed above, as the _SIP_ part might still be =
standard, at least for some methods etc. But still the communication =
would fail even if the tuple looks OK. (Also if I saw the =
characteristics above, I could as well determine that they describe =
normal VoIP application running in (very) some old PC with half-duplex =
audio drivers, so I claim the service description is hard to make =
unambiguous in the first place.) =20
 =20
I think the main point of a service tuple is to give the watcher enough =
information to know whether he can really communicate & interoperate =
with the advertised service. Given that many services taht are using SIP =
also have propritary or non-SIP features, I don't think the current =
approach is enough.=20

Of course each of these proprietary services are allowed to define =
additional status or tuple-level extensions to PIDF. However, I would =
like to have some kind of "service identifier" as part of the basic =
framework so that this could be done in a consistent manner. I think it =
would help especially in making of authorization and composition rules =
more simple. The current "class" attribute is way too loosely defined.

So the concrete proposal is to include in RPID a "service id" element =
that would have a vendor-specific namespace, similar to e.g. =
vendor-specific XCAP AUIDs. So for instance the SIP-based PTT app from =
vendor xyz.com would have, e.g.=20

<service-id>com.xyz.ptt</service-id>

while the PTT app compliant with OMA would have, e.g.

<service-id>com.openmobilealliance.ptt</service-id>

in _addition_ to describing sip:, half-duplex audio, and the supported =
methods. Now:
1.) Watcher having one of those apps could see right away whether it is =
possible to communicate
2.) Composer getting PUBLISHes from 2 sources could immediately know =
that they are from a similar app
3.) The app could set its authorization rules using the unique id.

The main downside I can see in this approach that if there are two =
different proprietary apps that indeed could communicate at least =
partially, the watcher might make a conclusion that communication is not =
possible baed on different service-id. However, in this case the watcher =
still would have the characteristics visible and could determine that =
some interworking could be achieved. For this kind of reasons there =
should be careful text about when to use service-id in the first place, =
and what can be determined from it.

Does someone think that this is an absolutely bad idea? If so, how would =
you envision solving the issues discussed above?

Regards,
	Markus=20


_______________________________________________
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From simple-bounces@ietf.org  Thu Sep 23 07:49:51 2004
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
To: "'simple@ietf.org'" <simple@ietf.org>
Subject: RE: [Simple] Presence Data Model: Identifying services
Date: Thu, 23 Sep 2004 13:41:37 +0200
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Hello,

Comments inline

P.S.: In general, I find the Presence Data model draft a good document. This
is how things should have started in the first place.

Greetings,
Stefan.

> -----Original Message-----
> From: simple-bounces@ietf.org 
> [mailto:simple-bounces@ietf.org] On Behalf Of Markus.Isomaki@nokia.com
> Sent: donderdag 23 september 2004 10:10
> To: simple@ietf.org
> Subject: [Simple] Presence Data Model: Identifying services
> 
> 
> Hi all,
> 
> I have a comment on Presence Data Model draft about the 
> identification/characterization of services. (I know this has 
> been discussed for a while already, but I still think the 
> current way is somewhat unadequate.)
> 
> The current Data Model draft says:
> 
>    In this model, services are not explicitly enumerated.  That is,
>    there is no "service" attribute with values of "ptt" or 
> "telephony".
>    Rather, the service is identified in one of two ways.  In 
> many cases,
>    the URI scheme is a clear indicator of service.  An "sms" 
> URI clearly
>    indicates SMS, and a "tel" URI clearly indicates 
> telephony.  For some
>    URIs, there may be many services available, for example, SIP.  For
>    those services, each service has a set of characteristics, each of
>    which has a well-defined meaning, such that a system can
>    unequivocally determine whether or not the service has that
>    characteristic.  This is discussed in more detail in [5]. 
> 
> I agree that the contact URI scheme is the most important 
> piece of information to distinguish what is the "service". 
> (There are some gaps here too, since e.g. the URI schemes for 
> SMS or MMS do not even exist, and there may be other non-IETF 
> protocols that face the same problem.) However, I'm not 
> convinced that listing the signaling/media characteristics of 
> the end-point or service really gives enough information to 
> the watcher to really determine if it can communicate with 
> the advertised service/application. 
> 
> The refenced draft [5] has a following example:
> 
> 5.6 Walkie-talkie
> 
>    The walkie-talkie service allows real-time voice communication
>    between participants.  Only one participant can speak at a 
> time; that
>    is, communication is half-duplex.  Typically, participants press a
>    button to indicate that they are ready to speak, although other
>    mechanism (e.g.  voice activation) are occasionally used.
> 
>    Support for the Walkie-Talkie service can be deduced by 
> observing the
>    presence of a contact URI with a scheme of "sip:", associated with
>    the following minimal set of capabilities: <audio>true</audio>
>    <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> 
> Presumably this is the same as the service sometimes called 
> Push-to-Talk (PTT, PoC). The concrete problem is that there 
> are several _non-interworking_ variants of this service, that 
> still all would match the characteristics listed above, as 
> the _SIP_ part might still be standard, at least for some 
> methods etc. But still the communication would fail even if 
> the tuple looks OK.
I don't think I understand you here. I would expect that if both end-point
SIP US support exactly the same methods, the SIP UA should also be able to
handle those SIP method. Or, to phase it in another way, both end-point
should be able to communicate with each other only based on using these
capabilities.

> (Also if I saw the characteristics above, 
> I could as well determine that they describe normal VoIP 
> application running in (very) some old PC with half-duplex 
> audio drivers, so I claim the service description is hard to 
> make unambiguous in the first place.)  
>
I also have the impression that it will be difficult to determine the
services merely based on the SIP methods (and ...) supported by the SIP UA.
It possibly might mean that the SIP-based ptt-service on my mobile phone
might be mapped onto some communication service on your PC. Then, I wonder
who will be responsible for such kind of mapping? Only the end terminals, or
is support from the network, presence server, necessary? And, how will one
service in one domain be mapped into a service in another domain? I believe
there are some open issues here. 
   
> I think the main point of a service tuple is to give the 
> watcher enough information to know whether he can really 
> communicate & interoperate with the advertised service. Given 
> that many services taht are using SIP also have propritary or 
> non-SIP features, I don't think the current approach is enough. 
> 
Do you have an example of a service that uses SIP and also has non
SIP-features? If your service also uses non-SIP features, you can not
determine the type of service by only looking at the SIP capabilities
supported by the SIP UA. Some indication of that non-SIP feature should also
be present in the presence document. 

> Of course each of these proprietary services are allowed to 
> define additional status or tuple-level extensions to PIDF.
Yes, you can always extend the PIDF. My comment here is that this might lead
to an "explosion" of applications or services. It will certainly not help
interoperability.
 
> However, I would like to have some kind of "service 
> identifier" as part of the basic framework so that this could 
> be done in a consistent manner. I think it would help 
> especially in making of authorization and composition rules 
> more simple. The current "class" attribute is way too loosely defined.
> 
> So the concrete proposal is to include in RPID a "service id" 
> element that would have a vendor-specific namespace, similar 
> to e.g. vendor-specific XCAP AUIDs. So for instance the 
> SIP-based PTT app from vendor xyz.com would have, e.g. 
> 
> <service-id>com.xyz.ptt</service-id>
> 
> while the PTT app compliant with OMA would have, e.g.
> 
> <service-id>com.openmobilealliance.ptt</service-id>
> 
In general, I could agree with that approach. However, I don't see a reason
why the vendor specific SIP-based PTT should be different from the OMA based
PTT. Simply define one PTT, based on the SIP UA capabilities.

> in _addition_ to describing sip:, half-duplex audio, and the 
> supported methods. Now:
> 1.) Watcher having one of those apps could see right away 
> whether it is possible to communicate
> 2.) Composer getting PUBLISHes from 2 sources could 
> immediately know that they are from a similar app
> 3.) The app could set its authorization rules using the unique id.
> 
> The main downside I can see in this approach that if there 
> are two different proprietary apps that indeed could 
> communicate at least partially, the watcher might make a 
> conclusion that communication is not possible baed on 
> different service-id. However, in this case the watcher still 
> would have the characteristics visible and could determine 
> that some interworking could be achieved. 
Personally, I don't think it is a good idea that an end-user should make
such decission. If I get a presence document that indicates that another
person is able to receive some type of communication, I expect that that
type of communication would actually work when I use it.

>For this kind of 
> reasons there should be careful text about when to use 
> service-id in the first place, and what can be determined from it.
> 
> Does someone think that this is an absolutely bad idea? If 
> so, how would you envision solving the issues discussed above?
> 
I think more care should be taken in mapping SIP UA capabilities into actual
services. That is not so clear to me!


> Regards,
> 	Markus 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

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Subject: RE: [Simple] Presence Data Model: Identifying services
Date: Thu, 23 Sep 2004 15:34:08 +0300
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Hi Stefan,

Comments back to you inline.=20

(Yes, I absolutely agree with you on the usefulness of the Presence Data =
Model draft and that it should have been the starting point, at least =
right after PIDF.)

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Goeman Stefan
> Sent: 23 September, 2004 14:42
> To: 'simple@ietf.org'
> Subject: RE: [Simple] Presence Data Model: Identifying services
>=20
>=20
> Hello,
>=20
> Comments inline
>=20
> P.S.: In general, I find the Presence Data model draft a good=20
> document. This
> is how things should have started in the first place.
>=20
> Greetings,
> Stefan.
>=20
> > -----Original Message-----
> > From: simple-bounces@ietf.org=20
> > [mailto:simple-bounces@ietf.org] On Behalf Of=20
> Markus.Isomaki@nokia.com
> > Sent: donderdag 23 september 2004 10:10
> > To: simple@ietf.org
> > Subject: [Simple] Presence Data Model: Identifying services
> >=20
> >=20
> > Hi all,
> >=20
> > I have a comment on Presence Data Model draft about the=20
> > identification/characterization of services. (I know this has=20
> > been discussed for a while already, but I still think the=20
> > current way is somewhat unadequate.)
> >=20
> > The current Data Model draft says:
> >=20
> >    In this model, services are not explicitly enumerated.  That is,
> >    there is no "service" attribute with values of "ptt" or=20
> > "telephony".
> >    Rather, the service is identified in one of two ways.  In=20
> > many cases,
> >    the URI scheme is a clear indicator of service.  An "sms"=20
> > URI clearly
> >    indicates SMS, and a "tel" URI clearly indicates=20
> > telephony.  For some
> >    URIs, there may be many services available, for example,=20
> SIP.  For
> >    those services, each service has a set of=20
> characteristics, each of
> >    which has a well-defined meaning, such that a system can
> >    unequivocally determine whether or not the service has that
> >    characteristic.  This is discussed in more detail in [5].=20
> >=20
> > I agree that the contact URI scheme is the most important=20
> > piece of information to distinguish what is the "service".=20
> > (There are some gaps here too, since e.g. the URI schemes for=20
> > SMS or MMS do not even exist, and there may be other non-IETF=20
> > protocols that face the same problem.) However, I'm not=20
> > convinced that listing the signaling/media characteristics of=20
> > the end-point or service really gives enough information to=20
> > the watcher to really determine if it can communicate with=20
> > the advertised service/application.=20
> >=20
> > The refenced draft [5] has a following example:
> >=20
> > 5.6 Walkie-talkie
> >=20
> >    The walkie-talkie service allows real-time voice communication
> >    between participants.  Only one participant can speak at a=20
> > time; that
> >    is, communication is half-duplex.  Typically,=20
> participants press a
> >    button to indicate that they are ready to speak, although other
> >    mechanism (e.g.  voice activation) are occasionally used.
> >=20
> >    Support for the Walkie-Talkie service can be deduced by=20
> > observing the
> >    presence of a contact URI with a scheme of "sip:",=20
> associated with
> >    the following minimal set of capabilities: <audio>true</audio>
> >    <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> >=20
> > Presumably this is the same as the service sometimes called=20
> > Push-to-Talk (PTT, PoC). The concrete problem is that there=20
> > are several _non-interworking_ variants of this service, that=20
> > still all would match the characteristics listed above, as=20
> > the _SIP_ part might still be standard, at least for some=20
> > methods etc. But still the communication would fail even if=20
> > the tuple looks OK.
> I don't think I understand you here. I would expect that if=20
> both end-point
> SIP US support exactly the same methods, the SIP UA should=20
> also be able to
> handle those SIP method. Or, to phase it in another way, both=20
> end-point
> should be able to communicate with each other only based on=20
> using these
> capabilities.
>

To some extent that might be possible, but there might be differences =
e.g. in floor control etc., which make the service still practically =
unusable, which means that the presence document shouldn't give the =
impression that the communication will work. Obviously it is possible to =
list all those additional characteristics as well (and this might be a =
good idea too), but for instance if presence of a network game is =
published, it is easier to _explicitely_ tell what the game is by =
providing some sort of unique id which would be only meaningful to those =
similar game clients. =20
=20
> > (Also if I saw the characteristics above,=20
> > I could as well determine that they describe normal VoIP=20
> > application running in (very) some old PC with half-duplex=20
> > audio drivers, so I claim the service description is hard to=20
> > make unambiguous in the first place.) =20
> >
> I also have the impression that it will be difficult to determine the
> services merely based on the SIP methods (and ...) supported=20
> by the SIP UA.
> It possibly might mean that the SIP-based ptt-service on my=20
> mobile phone
> might be mapped onto some communication service on your PC.=20
> Then, I wonder
> who will be responsible for such kind of mapping? Only the=20
> end terminals, or
> is support from the network, presence server, necessary? And,=20
> how will one
> service in one domain be mapped into a service in another=20
> domain? I believe
> there are some open issues here.=20
>

I think that it is fully OK to describe some rather simple things using =
characteristics, but in many cases applications are more complex.
   =20
> > I think the main point of a service tuple is to give the=20
> > watcher enough information to know whether he can really=20
> > communicate & interoperate with the advertised service. Given=20
> > that many services taht are using SIP also have propritary or=20
> > non-SIP features, I don't think the current approach is enough.=20
> >=20
> Do you have an example of a service that uses SIP and also has non
> SIP-features? If your service also uses non-SIP features, you can not
> determine the type of service by only looking at the SIP capabilities
> supported by the SIP UA. Some indication of that non-SIP=20
> feature should also
> be present in the presence document.=20
>=20

True, and since some of those non-SIP features might not be based on =
(IETF) standards, it might be good to have the possibility for the =
developers of such applications to have a short cut of saying what the =
app really is, rather than having to list characteristics. In the end, =
some applications are really not even meant to work with any other than =
the exactly same application, for instance some networking games.=20

> > Of course each of these proprietary services are allowed to=20
> > define additional status or tuple-level extensions to PIDF.
> Yes, you can always extend the PIDF. My comment here is that=20
> this might lead
> to an "explosion" of applications or services. It will=20
> certainly not help
> interoperability.
> =20

And who would standardize such characteristic definitions. It has =
already taken a couple of years to standardize even "prescaps", which is =
just a starting point.=20

> > However, I would like to have some kind of "service=20
> > identifier" as part of the basic framework so that this could=20
> > be done in a consistent manner. I think it would help=20
> > especially in making of authorization and composition rules=20
> > more simple. The current "class" attribute is way too=20
> loosely defined.
> >=20
> > So the concrete proposal is to include in RPID a "service id"=20
> > element that would have a vendor-specific namespace, similar=20
> > to e.g. vendor-specific XCAP AUIDs. So for instance the=20
> > SIP-based PTT app from vendor xyz.com would have, e.g.=20
> >=20
> > <service-id>com.xyz.ptt</service-id>
> >=20
> > while the PTT app compliant with OMA would have, e.g.
> >=20
> > <service-id>com.openmobilealliance.ptt</service-id>
> >=20
> In general, I could agree with that approach. However, I=20
> don't see a reason
> why the vendor specific SIP-based PTT should be different=20
> from the OMA based
> PTT. Simply define one PTT, based on the SIP UA capabilities.
>=20

Well, for instance Nokia has a prorietary PTT application, which uses =
SIP INVITE etc., so users are essentially addressed with sip: URI. =
However, it has some floor control, media etc. features which make it =
basically non-interoperable with clients that are not specifically =
following that proprietary spec.=20

OMA PTT is closer to standard IETF SIP. Unless there is some conversion, =
it does not work with Nokia PTT directly. There should be some way for =
distinguishing these two in a presence document, so that there won't be =
confusion. I think the current idea is to do some kind of proprietary =
PIDF extensions, but in my opinion a standard XML element in RPID to =
tell this kind of information would be better, also for the purposes of =
authorization rule making.

> > in _addition_ to describing sip:, half-duplex audio, and the=20
> > supported methods. Now:
> > 1.) Watcher having one of those apps could see right away=20
> > whether it is possible to communicate
> > 2.) Composer getting PUBLISHes from 2 sources could=20
> > immediately know that they are from a similar app
> > 3.) The app could set its authorization rules using the unique id.
> >=20
> > The main downside I can see in this approach that if there=20
> > are two different proprietary apps that indeed could=20
> > communicate at least partially, the watcher might make a=20
> > conclusion that communication is not possible baed on=20
> > different service-id. However, in this case the watcher still=20
> > would have the characteristics visible and could determine=20
> > that some interworking could be achieved.=20
> Personally, I don't think it is a good idea that an end-user=20
> should make
> such decission. If I get a presence document that indicates=20
> that another
> person is able to receive some type of communication, I=20
> expect that that
> type of communication would actually work when I use it.
>=20

Exactly right. If you are an application who does not necessarily aim =
for interoperability, but still uses sip:, INVITE etc., you should be =
able to explicitely say so.

> >For this kind of=20
> > reasons there should be careful text about when to use=20
> > service-id in the first place, and what can be determined from it.
> >=20
> > Does someone think that this is an absolutely bad idea? If=20
> > so, how would you envision solving the issues discussed above?
> >=20
> I think more care should be taken in mapping SIP UA=20
> capabilities into actual
> services. That is not so clear to me!
>

The idea is nice, but I don't see a problem in _addition_ defining the =
application with something like "com.openmobilealliance.ptt" or =
"com.nokia.ptt", if that kind of profile would have more meaning than =
the list of characteristics.
=20
>=20
> > Regards,
> > 	Markus=20
> >=20
> >=20
> > _______________________________________________
> > Simple mailing list
> > Simple@ietf.org
> > https://www1.ietf.org/mailman/listinfo/simple
> >=20
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

Markus

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From simple-bounces@ietf.org  Thu Sep 23 08:59:45 2004
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From: "Pete Cordell" <pete@tech-know-ware.com>
To: <hisham.khartabil@nokia.com>
References: <5816828233DEFA41807A6CFDFDF2343C08B62E@esebe056.ntc.nokia.com>
Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Thu, 23 Sep 2004 13:55:32 +0100
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Hi Hisham,

For my money, your statements are roughly right.  The only concern I have is
the bit where it says: "...consumer MUST validate...".  It's probably not
the intent, but there's a potential implication that the IETF is saying that
consumers must run a separate validation exercise on the received data
(rather than say implicitly validating while the application interprets the
data, in which case they may only end up validating the parts of the data
they are interested in).  It sounds like the IETF is saying how people
should build their devices which traditionally they haven't done.  It's
probably not a big issue, but a possible re-wording is:-

"The XML document received by a consumer MUST be valid according the
schemas, including extension schemas, the consumer has access to that are
applicable to this XML document.  The consumer MAY validate the received XML
document against such schemas, and MAY reject the XML document if such
validation is not successful."

I'll leave it to you.

Pete.
--
=============================================
Pete Cordell
Tech-Know-Ware
pete(at)tech-know-ware.com
http://www.tech-know-ware.com
http://www.xml2cpp.com - Now Available
+44 1473 635863
=============================================

----- Original Message ----- 
From: <hisham.khartabil@nokia.com>
To: <Pekka.Pessi@nokia.com>; <jdrosen@dynamicsoft.com>
Cc: <Pekka.Pessi@nokia.com>; <Jari.Urpalainen@nokia.com>; <simple@ietf.org>;
<RjS@xten.com>
Sent: Wednesday, September 22, 2004 9:54 AM
Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE
documents


How does this proposal sound (for the filter drafts):

In the format document, the following text appears: The XML document MUST be
well-formed and MUST be valid according to schemas, including extensions
schemas, available to the validater and applicable to the XML document.

In the functionality draft: The creator MUST ensure that the XML document is
well-formed and valid. The creator MUST NOT insert any extension elements or
attributes into the XML document unless it has access to the extension
schema and can validate the XML document. The XML document consumer MUST
validate the XML document according the schemas, including extension
schemas, it has access to that are applicable to this XML document.

For pidf, the second part might be slightly different since the creator of
the document can be the compositor and the compositor can create a presence
XML document by aggregating tuples that contain extensions that the
compositor is unaware of. Perhaps something can go in the Date Model draft
discussing this.

Regards,
Hisham



> -----Original Message-----
> From: simple-bounces@ietf.org
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Pekka Pessi
> Sent: 20.September.2004 14:36
> To: Jonathan Rosenberg
> Cc: Pessi Pekka (Nokia-NRC/Helsinki); Urpalainen Jari
> (Nokia-NRC/Helsinki); simple@ietf.org; Robert Sparks
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>
>
> Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> >I'm saying that a document of type application/pidf+xml
> could be valid
> >against the PIDF schema, but it contains extensions (which
> are allowed in
> >the pidf schema), and those extensions make use of elements
> defined in
> >another schema. However, those elements are not valid
> according to that
> >other schema.
>
> >In this case, do we consider this document "valid"?
>
> Term valid has at least three different senses with the XML:
>
> 1)
>
> W3C defines that an XML document is valid when it has a DTD and
> the document complies with the constraints set forth in DTD.
>
> That is not what we want. No <!DOCTYPE> for presence, no "MUST be
> valid", please.
>
> (When RFC 3863 says that the application/pidf+xml document MUST
> be well-formed and SHOULD be valid, I guess it just says that
> extensions are allowed, even if they are not shown in the DTD.)
>
> 2)
>
> XML schema specification uses a bit different terminology. Schemas
> don't specify documents, but trees (item and all its descendants).
> You can assess validity of a particular tree, for instance, all
> the XML stuff within <presence> element. The validator can also
> skip certain elements (like the wildcard elements used for
> extending pidf). So, validator can return different results: tree
> and items within it are fully validated, partially validated, or
> not validated at all. Also each, item within the tree are valid or
> invalid or their validity is not known.
>
> I'd say that the XML document is schema-valid if its root element
> is fully validated and it and all its descendants are valid. This
> is not the official definition, however.
>
> This is unfeasible for the intermediate nodes (or processes), too.
> Unless the node knows the schema (and semantics) of all the
> extensions, it cannot assess the validity of composed or filtered
> documents. I think filtering and composing is precisely the things
> that we want to do in the server nodes.
>
> I can also imagine some applications that would just use existing
> presence service, just add their non-standard application-specific
> status in the published presence document. Now imagine two of
> those applications running on the same node.
>
> 3)
>
> Beside validity assessment, XML schema specification has the
> concept of "local schema-validity", too:
>
> <http://www.w3.org/TR/xmlschema-1/#section-Overview-of-XML-Schema>
>
> I must admit that I don't fully understand what this
> means, but I have a good hunch that it is what most of us
> want. ;)
>
> A element or attribute is valid if it satisfies the constraints
> set by the XML schema. So, <presence> element is valid, if it
> contains entity attribute, zero or more <tuple> elements, then
> zero or more <note> elements and after then something wildcarded
> stuff (I guess that is anything else but pidf-namespace stuff). I
> don't understand how the datatype checking of evil types like
> xs:ID or xs:IDREF is done when the "local schema-validity" is
> determined. I *guess* that ID/IDREF attribute in tuple is valid if
> its value is syntactically correct. Like id="_123" is valid but,
> id="123" is not. No check for all id's within a document/item and
> all of its descendats is required.
>
> So, if we have a application/pidf+xml document, we can simply say
> that it MUST conform to its schema and other requirements set
> forth in RFC 3863. If we want to be more verbose, we could say
> that the root node in the document MUST be partially or fully
> validated and none of the validated nodes MUST NOT be invalid. Or
> we could invent our own term and say that documents MUST be
> locally-schema-valid with the schema XXX.
>
> --Pekka
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>

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From simple-bounces@ietf.org  Thu Sep 23 12:16:15 2004
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To: Markus.Isomaki@nokia.com
Subject: Re: [Simple] Presence Data Model: Identifying services
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I have comments both to Markus' original posting and to the ensuing 
discussion. So I will just insert inline in the latest response I have.

	Paul

Markus.Isomaki@nokia.com wrote:
> Hi Stefan,
> 
> Comments back to you inline. 
> 
> (Yes, I absolutely agree with you on the usefulness of the Presence Data Model draft and that it should have been the starting point, at least right after PIDF.)
> 
> 
>>-----Original Message-----
>>From: simple-bounces@ietf.org 
>>[mailto:simple-bounces@ietf.org]On Behalf
>>Of ext Goeman Stefan
>>Sent: 23 September, 2004 14:42
>>To: 'simple@ietf.org'
>>Subject: RE: [Simple] Presence Data Model: Identifying services
>>
>>
>>Hello,
>>
>>Comments inline
>>
>>P.S.: In general, I find the Presence Data model draft a good 
>>document. This
>>is how things should have started in the first place.
>>
>>Greetings,
>>Stefan.
>>
>>
>>>-----Original Message-----
>>>From: simple-bounces@ietf.org 
>>>[mailto:simple-bounces@ietf.org] On Behalf Of 
>>
>>Markus.Isomaki@nokia.com
>>
>>>Sent: donderdag 23 september 2004 10:10
>>>To: simple@ietf.org
>>>Subject: [Simple] Presence Data Model: Identifying services
>>>
>>>
>>>Hi all,
>>>
>>>I have a comment on Presence Data Model draft about the 
>>>identification/characterization of services. (I know this has 
>>>been discussed for a while already, but I still think the 
>>>current way is somewhat unadequate.)
>>>
>>>The current Data Model draft says:
>>>
>>>   In this model, services are not explicitly enumerated.  That is,
>>>   there is no "service" attribute with values of "ptt" or 
>>>"telephony".
>>>   Rather, the service is identified in one of two ways.  In 
>>>many cases,
>>>   the URI scheme is a clear indicator of service.  An "sms" 
>>>URI clearly
>>>   indicates SMS, and a "tel" URI clearly indicates 
>>>telephony. 

I don't find the scheme as clearcut as that. I don't think one should 
assume "tel" indicates telephony. It perhaps does imply reachability via 
various telephony protocols for purpose of voice communications, but is 
not exclusive to that. If the tel: URI can be mapped to a sip endpoint 
and protocol, then non-voice, non-telephony capabilities are not excluded.

For instance, it may prove convenient to publish a tel URI for a sip 
endpoint that is capable of both voice and IM. Then calls via the PSTN 
will work, calls from a sip audio phone will work, calls from another 
voice+im client will have access to voice and IM.

 >>> For some
>>>   URIs, there may be many services available, for example, 
>>
>>SIP.  For
>>
>>>   those services, each service has a set of 
>>
>>characteristics, each of
>>
>>>   which has a well-defined meaning, such that a system can
>>>   unequivocally determine whether or not the service has that
>>>   characteristic.  This is discussed in more detail in [5]. 
>>>
>>>I agree that the contact URI scheme is the most important 
>>>piece of information to distinguish what is the "service". 
>>>(There are some gaps here too, since e.g. the URI schemes for 
>>>SMS or MMS do not even exist, and there may be other non-IETF 
>>>protocols that face the same problem.) However, I'm not 
>>>convinced that listing the signaling/media characteristics of 
>>>the end-point or service really gives enough information to 
>>>the watcher to really determine if it can communicate with 
>>>the advertised service/application. 
>>>
>>>The refenced draft [5] has a following example:
>>>
>>>5.6 Walkie-talkie
>>>
>>>   The walkie-talkie service allows real-time voice communication
>>>   between participants.  Only one participant can speak at a 
>>>time; that
>>>   is, communication is half-duplex.  Typically, 
>>
>>participants press a
>>
>>>   button to indicate that they are ready to speak, although other
>>>   mechanism (e.g.  voice activation) are occasionally used.
>>>
>>>   Support for the Walkie-Talkie service can be deduced by 
>>>observing the
>>>   presence of a contact URI with a scheme of "sip:", 
>>
>>associated with
>>
>>>   the following minimal set of capabilities: <audio>true</audio>
>>>   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>>
>>>Presumably this is the same as the service sometimes called 
>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
>>>are several _non-interworking_ variants of this service, that 
>>>still all would match the characteristics listed above, as 
>>>the _SIP_ part might still be standard, at least for some 
>>>methods etc. But still the communication would fail even if 
>>>the tuple looks OK.
>>
>>I don't think I understand you here. I would expect that if 
>>both end-point
>>SIP US support exactly the same methods, the SIP UA should 
>>also be able to
>>handle those SIP method. Or, to phase it in another way, both 
>>end-point
>>should be able to communicate with each other only based on 
>>using these
>>capabilities.
>>
> 
> 
> To some extent that might be possible, but there might be differences e.g. in floor control etc., which make the service still practically unusable, which means that the presence document shouldn't give the impression that the communication will work. Obviously it is possible to list all those additional characteristics as well (and this might be a good idea too), but for instance if presence of a network game is published, it is easier to _explicitely_ tell what the game is by providing some sort of unique id which would be only meaningful to those similar game clients.
 >
>>>(Also if I saw the characteristics above, 
>>>I could as well determine that they describe normal VoIP 
>>>application running in (very) some old PC with half-duplex 
>>>audio drivers, so I claim the service description is hard to 
>>>make unambiguous in the first place.)  

I think the goal should be that all of these are at least minimally 
interoperable. I ought to be able to call a PTT contact from a regular 
voice phone and have *something* reasonable happen. (Maybe it is just 
oneway voice, with the non-ptt caller never getting the floor, but that 
may be better than nothing.

I think the real problem in this case is not with the capability 
approach. It is that the 'duplex' capability is underspecified. There is 
no explanation of how a pair of half duplex endpoints (or one 
half-duplex and one full-duplex endpoint) agree on who can talk.

In the case of the 'voice' capability, it is understood that having both 
endpoints support voice isn't enough to support communication. There is 
an SDP negotiation of codecs, etc. that either succeeds or fails. There 
should be something similar for half-duplex.

>>I also have the impression that it will be difficult to determine the
>>services merely based on the SIP methods (and ...) supported 
>>by the SIP UA.
>>It possibly might mean that the SIP-based ptt-service on my 
>>mobile phone
>>might be mapped onto some communication service on your PC. 
>>Then, I wonder
>>who will be responsible for such kind of mapping? Only the 
>>end terminals, or
>>is support from the network, presence server, necessary? And, 
>>how will one
>>service in one domain be mapped into a service in another 
>>domain? I believe
>>there are some open issues here. 

The intent is that capabilities have common meanings in all domains, so 
they don't need to be mapped. I don't see why there should be any 
binding between services and domains.

In the general case one should assume that there are devices of widely 
varying capabilities in every domain, and that they know best what they 
are capable of. So the device should be the one deciding if a contact 
with and advertised set of capabilties is something it wants to attempt 
communication with.

In some environment where there is an intermediary in front of a device 
that understands the device and acts on its behalf, then I suppose it 
could make these decisions. But I would consider that the odd case.

> I think that it is fully OK to describe some rather simple things using characteristics, but in many cases applications are more complex.

Clearly we have chosen not to describe capabilities if full detail. 
(E.g. we don't publish codec support.) Because of this, there is the 
possibility that we make a wrong decision about ability to communicate 
and it doesn't get discovered until we try.

This is ok when the expected probability of failure is low. (We 
generally expect that there will be some codec in common, or that one 
end or the other will be able to introduce a transcoder.)

>>>I think the main point of a service tuple is to give the 
>>>watcher enough information to know whether he can really 
>>>communicate & interoperate with the advertised service. Given 
>>>that many services taht are using SIP also have propritary or 
>>>non-SIP features, I don't think the current approach is enough. 

In cases where the existing capabilities aren't sufficient to ensure a 
good probability of communication, then we should indeed consider adding 
additional capabilities.

But I continue to object to the notion of "service" as the way to do 
that, because it conveys no semantics about what the capability is that 
it describes.

>>Do you have an example of a service that uses SIP and also has non
>>SIP-features? If your service also uses non-SIP features, you can not
>>determine the type of service by only looking at the SIP capabilities
>>supported by the SIP UA. Some indication of that non-SIP 
>>feature should also
>>be present in the presence document. 
>>
> 
> 
> True, and since some of those non-SIP features might not be based on (IETF) standards, it might be good to have the possibility for the developers of such applications to have a short cut of saying what the app really is, rather than having to list characteristics. In the end, some applications are really not even meant to work with any other than the exactly same application, for instance some networking games. 

If the intent is to define a closed world, where interoperability is not 
even considered a desirable feature, then I agree. I think that may 
indeed be the case for proprietary games.

But for something like PTT, while it might be in the best interest of 
the maker of a particular PTT solution to keep it closed, I don't think 
it is in the best interest of society or the IETF that this happen. This 
is precisely the forum in which we try to prevent that from happening.

>>>Of course each of these proprietary services are allowed to 
>>>define additional status or tuple-level extensions to PIDF.
>>
>>Yes, you can always extend the PIDF. My comment here is that 
>>this might lead
>>to an "explosion" of applications or services. It will 
>>certainly not help
>>interoperability.

My point precisely.

> And who would standardize such characteristic definitions. It has already taken a couple of years to standardize even "prescaps", which is just a starting point. 
> 
> 
>>>However, I would like to have some kind of "service 
>>>identifier" as part of the basic framework so that this could 
>>>be done in a consistent manner. I think it would help 
>>>especially in making of authorization and composition rules 
>>>more simple. The current "class" attribute is way too 
>>loosely defined.
>>
>>>So the concrete proposal is to include in RPID a "service id" 
>>>element that would have a vendor-specific namespace, similar 
>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
>>><service-id>com.xyz.ptt</service-id>
>>>while the PTT app compliant with OMA would have, e.g.
>>><service-id>com.openmobilealliance.ptt</service-id>

I am not fond of this solution. Done this way, we have defined an 
element that has no semantics at all, only syntax.

I would rather see vendor specific extensions to PIDF. At least then the 
people that support an extension will understand what it means.

But I would rather see people come forward and do the hard work of 
publicly defining new capabilities so that independent implementations 
can use them in an interoperable way.

>>In general, I could agree with that approach. However, I 
>>don't see a reason
>>why the vendor specific SIP-based PTT should be different 
>>from the OMA based
>>PTT. Simply define one PTT, based on the SIP UA capabilities.
>>
> 
> 
> Well, for instance Nokia has a prorietary PTT application, which uses SIP INVITE etc., so users are essentially addressed with sip: URI. However, it has some floor control, media etc. features which make it basically non-interoperable with clients that are not specifically following that proprietary spec. 

This is a problem, not a good thing. I realize it has to happen as part 
of the development process, but the goal should be to get out of it as 
fast as possible.

I would hope that endpoints supporting this would be described as well 
as possible using existing capabilities, and that a reasonable effort be 
made to do something reasonable when some other sip device connects.

If you really can't deal with a regular voice caller at all, then I 
would suggest, as a transitional step, saying that you don't support 
voice, and then advertising some proprietary capability as an 
alternative to voice.

> OMA PTT is closer to standard IETF SIP. Unless there is some conversion, it does not work with Nokia PTT directly. There should be some way for distinguishing these two in a presence document, so that there won't be confusion. I think the current idea is to do some kind of proprietary PIDF extensions, but in my opinion a standard XML element in RPID to tell this kind of information would be better, also for the purposes of authorization rule making.

See above for what I think you could do for now. The same issues may 
apply to OMA PTT, and a similar solution could be applied to it.

Hopefully somebody will come up with a PTT that can be dealt with more 
gracefully. I think that work should be done on better defining 
half-duplex, so that it can be used in conjunction with voice to 
describe PTT. But it may just turn out to be the wrong thing. Maybe we 
need a "floor-control" capability.

>>>in _addition_ to describing sip:, half-duplex audio, and the 
>>>supported methods. Now:
>>>1.) Watcher having one of those apps could see right away 
>>>whether it is possible to communicate
>>>2.) Composer getting PUBLISHes from 2 sources could 
>>>immediately know that they are from a similar app
>>>3.) The app could set its authorization rules using the unique id.
>>>
>>>The main downside I can see in this approach that if there 
>>>are two different proprietary apps that indeed could 
>>>communicate at least partially, the watcher might make a 
>>>conclusion that communication is not possible baed on 
>>>different service-id.

This is my concern as well.

 >>> However, in this case the watcher still
>>>would have the characteristics visible and could determine 
>>>that some interworking could be achieved. 
>>
>>Personally, I don't think it is a good idea that an end-user 
>>should make
>>such decission. If I get a presence document that indicates 
>>that another
>>person is able to receive some type of communication, I 
>>expect that that
>>type of communication would actually work when I use it.

With high probability. There are no guarantees with presence. For 
instance see my mention above of codecs.

> Exactly right. If you are an application who does not necessarily aim for interoperability, but still uses sip:, INVITE etc., you should be able to explicitely say so.

And you can, without a special notion of service.

>>>For this kind of 
>>>reasons there should be careful text about when to use 
>>>service-id in the first place, and what can be determined from it.
>>>
>>>Does someone think that this is an absolutely bad idea? If 
>>>so, how would you envision solving the issues discussed above?

I still don't buy this.

>>I think more care should be taken in mapping SIP UA 
>>capabilities into actual
>>services. That is not so clear to me!
> 
> The idea is nice, but I don't see a problem in _addition_ defining the application with something like "com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind of profile would have more meaning than the list of characteristics.

I am still opposed to defining an attribute with no semantics.
We have been around this so many times that I am quite certain that it 
will never be possible to get a useful agreement on what "service" means.

If it is introduced, I expect it to have problems like those of the INFO 
method. (The arguments for introducing it seem similar.)

	Paul


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From: Goeman Stefan <Stefan.Goeman@siemens.com>
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Hello All,

I have been reading the draft on Presence Authorization Rules, and also the
draft draft-ietf-geopriv-common-policy.
I have some remarks (otherwise I would not mail of course).

First of all, the draft on Presence Authorization Rules contracts the
geopriv draft, at least in one place.
In Section 3.2 on Actions, values are given to block, confirm, ... . If you
have two rules for the same entity and you combine them according to the
combining rules as explained in the geopriv draft you end up with the
highest value. Yet, in this case the highest value is the least privacy
preserving. This is in contradiction with the statement on page 21 of the
geopriv draft:
"Furthermore, permissions and the meaning of their values MUST be defined in
such a way that the usage of combining rules CR1, CR2, and CR3 always
preserves or increases the level of privacy protection for the PT". 

One can argue what is wrong here. Personally, I think the statement in the
geopriv draft will not really work well (see also below).

Additionally, wouldn't it also be nice if one could group people into a
class of people and define a rule for that class of people. For example, you
could have a class of friends, family, colleagues and define a rule each of
this class. The problem know arises that one person could belong to
different groups. Then again, the statement in the geopriv draft will not
work well. 
But, the best example I could think of would be the following. Suppose you
have a rule for your friends (bob@example.com belong to this group) and an
separate rule for bob@example.com. Now, why would you create a separate rule
for bob@example.com. I would say to grant him more privileges, not less. I
believe it will be easier to provide limited privileges to groups, and more
privileges to "the happy few". Certainly in this case, the geopriv statement
will not work; it has the opposite effect.

So, In general I would say that the combining rules should be formulated in
a way that it decreases the privacy of the presentity. This comes also from
the fact that it was chosen to use positive rules! The same functionality
could be reached by using negative rules. 


Greetings,
Stefan.

Stefan Goeman 
SIEMENS ICM/ICN
ICM N D SE 
Stefan.Goeman@siemens.com  <mailto:Stefan.Goeman@siemens.atea.be> 
phone:   +32 14 253020 
mobile:  
Fax:       +32 14 22 29 94 


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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2657.73">
<TITLE>Presence Authorization Rules</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hello All,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I have been reading the draft on =
Presence Authorization Rules, and also the draft =
draft-ietf-geopriv-common-policy.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">I have some remarks (otherwise I =
would not mail of course).</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">First of all, the draft on Presence =
Authorization Rules contracts the geopriv draft, at least in one =
place.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">In Section 3.2 on Actions, values are =
given to block, confirm, ... . If you have two rules for the same =
entity and you combine them according to the combining rules as =
explained in the geopriv draft you end up with the highest value. Yet, =
in this case the highest value is the least privacy preserving. This is =
in contradiction with the statement on page 21 of the geopriv =
draft:</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&quot;Furthermore, permissions and the =
meaning of their values MUST be defined in such a way that the usage of =
combining rules CR1, CR2, and CR3 always preserves or increases the =
level of privacy protection for the PT&quot;. </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">One can argue what is wrong here. =
Personally, I think the statement in the geopriv draft will not really =
work well (see also below).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Additionally, wouldn't it also be nice =
if one could group people into a class of people and define a rule for =
that class of people. For example, you could have a class of friends, =
family, colleagues and define a rule each of this class. The problem =
know arises that one person could belong to different groups. Then =
again, the statement in the geopriv draft will not work well. =
</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">But, the best example I could think of =
would be the following. Suppose you have a rule for your friends =
(bob@example.com belong to this group) and an separate rule for =
bob@example.com. Now, why would you create a separate rule for =
bob@example.com. I would say to grant him more privileges, not less. I =
believe it will be easier to provide limited privileges to groups, and =
more privileges to &quot;the happy few&quot;. Certainly in this case, =
the geopriv statement will not work; it has the opposite =
effect.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">So, In general I would say that the =
combining rules should be formulated in a way that it decreases the =
privacy of the presentity. This comes also from the fact that it was =
chosen to use positive rules! The same functionality could be reached =
by using negative rules. </FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Greetings,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Stefan.</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Stefan =
Goeman</FONT><BR>
<B></B><B><FONT COLOR=3D"#008080" FACE=3D"Times New =
Roman">SIEMENS</FONT><FONT FACE=3D"Arial"> </FONT><FONT =
COLOR=3D"#008080" SIZE=3D2 FACE=3D"Arial">ICM/ICN</FONT></B>
<BR><B><FONT COLOR=3D"#008080" SIZE=3D2 FACE=3D"Arial">ICM N D =
SE</FONT><FONT FACE=3D"Arial"></FONT></B><BR>
<A HREF=3D"mailto:Stefan.Goeman@siemens.atea.be"><U></U><U><FONT =
COLOR=3D"#0000FF" SIZE=3D1 =
FACE=3D"Verdana">Stefan.Goeman@siemens.com</FONT></U></A><BR>
<U></U><U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">phone</FONT></U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">:=A0=A0 +32 14 253020<BR>
</FONT><U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">mobile</FONT></U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">:=A0<BR>
</FONT><U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">Fax:</FONT></U><FONT COLOR=3D"#000000" SIZE=3D1 =
FACE=3D"Arial">=A0=A0=A0=A0=A0=A0 +32 14 22 29 94</FONT>=20
</P>

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From: Tschofenig Hannes <hannes.tschofenig@siemens.com>
To: "'Goeman Stefan'" <Stefan.Goeman@siemens.com>,
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Subject: RE: [Simple] Presence Authorization Rules
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# hi stefan, 
 
# please see my comments inline.

Hello All, 

I have been reading the draft on Presence Authorization Rules, and also the
draft draft-ietf-geopriv-common-policy. 
I have some remarks (otherwise I would not mail of course). 

First of all, the draft on Presence Authorization Rules contracts the
geopriv draft, at least in one place. 
In Section 3.2 on Actions, values are given to block, confirm, ... . If you
have two rules for the same entity and you combine them according to the
combining rules as explained in the geopriv draft you end up with the 
highest value.
# to be more precise the combining permissions algorithms is applicable if
two or more rules fire. 

 Yet, in this case the highest value is the least privacy preserving. This
is in contradiction with the statement on page 21 of the geopriv draft:

"Furthermore, permissions and the meaning of their values MUST be defined in
such a way that the usage of combining rules CR1, CR2, and CR3 always
preserves or increases the level of privacy protection for the PT". 

One can argue what is wrong here. Personally, I think the statement in the
geopriv draft will not really work well (see also below).

# what would happen if jonathan changes the values corresponding to the
listed actions from

Block (0)
Confirm (1)
Polite-block (2)
Allow (3)

# to the following values: 

Block (3)
Confirm (2)
Polite-block (1)
Allow (0)

# now assume that one rule with allow(0) and another one with block(3)
fires. the combining permissions algorithm would result in a block(3)
action. 

# are you more happy with this result? 


Additionally, wouldn't it also be nice if one could group people into a
class of people and define a rule for that class of people. For example, you
could have a class of friends, family, colleagues and define a rule each of
this class. The problem know arises that one person could belong to
different groups. Then again, the statement in the geopriv draft will not
work well. 
# grouping people together can be solved on a user interface basis. There is
not necessarily a need to specify groups in the policies itself (except if
you want to reduce the bandwidth consumption). 

~snip~

# ciao
# hannes

_______________________________________________
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Subject: RE: [Simple] Presence Data Model: Identifying services
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Hi Paul,

Thanks for your comments.=20

I guess what we both want to accomplish is this:

Allow the watcher to better understand whether and how it is able to =
communicate with a "service" or "communication mean" that is advertised =
by the presentity.=20

The current approach is to list as many characteristics and parameters =
of the service as possible, so that the watcher has as much information =
as possible. In theory this works well, as long as there is some agreed =
set of characteristics for each possible service. In practice there are =
no guarantees, since some essential info, such as codec support, is =
probably discovered only by trial and error. The concept still works =
well as long as the probability of a "false positive" is reasonably low.

One idea might be that the presentity should be able to declare some =
capabilities as mandatory/essential, i.e. saying like "if you don't =
support at least capabilities A and B, please don't contact me, since =
there is no value in this from my point of view". An example might be =
PTT, where the presentity could declare support for "voice", =
"half-duplex", "INVITE, ..." and floor control "foo", and might insist =
that even if the watcher supported the same except floor control "bar", =
the communication should not be attempted. Obviously it would be in the =
power of the clients and users to decide what to declare as essential =
and whether to obey those declarations, but at least it would give a =
hint whether things would work in a reasonable way or not. The ability =
to declare something as mandatory/essential should apply also to =
characteristics not understood by the watcher, e.g. if the watcher in =
the example is a normal VoIP client able to do half-duplex but with no =
understanding of any type of floor control (except maybe on layer 9 ;-), =
it could still see that "some feature I don't understand seems to be =
mandatory, so there is a high probability that the communication will =
not work". In this way the evil proprietary client could hint "please =
don't call me even though I'm addressable with SIP unless you know what =
xyz and zyx are, since if you don't, the outcome may not be anything =
useful".

Is there any chance of adding this kind of mechanism? I guess it would =
be some kind of attribute that would apply to all elements describing =
service characteristics. I believe it should be part of the data model, =
so that the designers of the proprietary extensions would know how to =
use it.=20

The hard part I see is keeping the standardized characteristics =
up-to-date and useful. For instance adding "floor control protocol" etc.

Markus


> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 23 September, 2004 18:47
> To: Isomaki Markus (Nokia-TP/Espoo)
> Cc: Stefan.Goeman@siemens.com; simple@ietf.org
> Subject: Re: [Simple] Presence Data Model: Identifying services
>=20
>=20
> I have comments both to Markus' original posting and to the ensuing=20
> discussion. So I will just insert inline in the latest=20
> response I have.
>=20
> 	Paul
>=20
> Markus.Isomaki@nokia.com wrote:
> > Hi Stefan,
> >=20
> > Comments back to you inline.=20
> >=20
> > (Yes, I absolutely agree with you on the usefulness of the=20
> Presence Data Model draft and that it should have been the=20
> starting point, at least right after PIDF.)
> >=20
> >=20
> >>-----Original Message-----
> >>From: simple-bounces@ietf.org=20
> >>[mailto:simple-bounces@ietf.org]On Behalf
> >>Of ext Goeman Stefan
> >>Sent: 23 September, 2004 14:42
> >>To: 'simple@ietf.org'
> >>Subject: RE: [Simple] Presence Data Model: Identifying services
> >>
> >>
> >>Hello,
> >>
> >>Comments inline
> >>
> >>P.S.: In general, I find the Presence Data model draft a good=20
> >>document. This
> >>is how things should have started in the first place.
> >>
> >>Greetings,
> >>Stefan.
> >>
> >>
> >>>-----Original Message-----
> >>>From: simple-bounces@ietf.org=20
> >>>[mailto:simple-bounces@ietf.org] On Behalf Of=20
> >>
> >>Markus.Isomaki@nokia.com
> >>
> >>>Sent: donderdag 23 september 2004 10:10
> >>>To: simple@ietf.org
> >>>Subject: [Simple] Presence Data Model: Identifying services
> >>>
> >>>
> >>>Hi all,
> >>>
> >>>I have a comment on Presence Data Model draft about the=20
> >>>identification/characterization of services. (I know this has=20
> >>>been discussed for a while already, but I still think the=20
> >>>current way is somewhat unadequate.)
> >>>
> >>>The current Data Model draft says:
> >>>
> >>>   In this model, services are not explicitly enumerated.  That is,
> >>>   there is no "service" attribute with values of "ptt" or=20
> >>>"telephony".
> >>>   Rather, the service is identified in one of two ways.  In=20
> >>>many cases,
> >>>   the URI scheme is a clear indicator of service.  An "sms"=20
> >>>URI clearly
> >>>   indicates SMS, and a "tel" URI clearly indicates=20
> >>>telephony.=20
>=20
> I don't find the scheme as clearcut as that. I don't think one should=20
> assume "tel" indicates telephony. It perhaps does imply=20
> reachability via=20
> various telephony protocols for purpose of voice=20
> communications, but is=20
> not exclusive to that. If the tel: URI can be mapped to a sip=20
> endpoint=20
> and protocol, then non-voice, non-telephony capabilities are=20
> not excluded.
>=20
> For instance, it may prove convenient to publish a tel URI for a sip=20
> endpoint that is capable of both voice and IM. Then calls via=20
> the PSTN=20
> will work, calls from a sip audio phone will work, calls from another=20
> voice+im client will have access to voice and IM.
>=20
>  >>> For some
> >>>   URIs, there may be many services available, for example,=20
> >>
> >>SIP.  For
> >>
> >>>   those services, each service has a set of=20
> >>
> >>characteristics, each of
> >>
> >>>   which has a well-defined meaning, such that a system can
> >>>   unequivocally determine whether or not the service has that
> >>>   characteristic.  This is discussed in more detail in [5].=20
> >>>
> >>>I agree that the contact URI scheme is the most important=20
> >>>piece of information to distinguish what is the "service".=20
> >>>(There are some gaps here too, since e.g. the URI schemes for=20
> >>>SMS or MMS do not even exist, and there may be other non-IETF=20
> >>>protocols that face the same problem.) However, I'm not=20
> >>>convinced that listing the signaling/media characteristics of=20
> >>>the end-point or service really gives enough information to=20
> >>>the watcher to really determine if it can communicate with=20
> >>>the advertised service/application.=20
> >>>
> >>>The refenced draft [5] has a following example:
> >>>
> >>>5.6 Walkie-talkie
> >>>
> >>>   The walkie-talkie service allows real-time voice communication
> >>>   between participants.  Only one participant can speak at a=20
> >>>time; that
> >>>   is, communication is half-duplex.  Typically,=20
> >>
> >>participants press a
> >>
> >>>   button to indicate that they are ready to speak, although other
> >>>   mechanism (e.g.  voice activation) are occasionally used.
> >>>
> >>>   Support for the Walkie-Talkie service can be deduced by=20
> >>>observing the
> >>>   presence of a contact URI with a scheme of "sip:",=20
> >>
> >>associated with
> >>
> >>>   the following minimal set of capabilities: <audio>true</audio>
> >>>   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> >>>
> >>>Presumably this is the same as the service sometimes called=20
> >>>Push-to-Talk (PTT, PoC). The concrete problem is that there=20
> >>>are several _non-interworking_ variants of this service, that=20
> >>>still all would match the characteristics listed above, as=20
> >>>the _SIP_ part might still be standard, at least for some=20
> >>>methods etc. But still the communication would fail even if=20
> >>>the tuple looks OK.
> >>
> >>I don't think I understand you here. I would expect that if=20
> >>both end-point
> >>SIP US support exactly the same methods, the SIP UA should=20
> >>also be able to
> >>handle those SIP method. Or, to phase it in another way, both=20
> >>end-point
> >>should be able to communicate with each other only based on=20
> >>using these
> >>capabilities.
> >>
> >=20
> >=20
> > To some extent that might be possible, but there might be=20
> differences e.g. in floor control etc., which make the=20
> service still practically unusable, which means that the=20
> presence document shouldn't give the impression that the=20
> communication will work. Obviously it is possible to list all=20
> those additional characteristics as well (and this might be a=20
> good idea too), but for instance if presence of a network=20
> game is published, it is easier to _explicitely_ tell what=20
> the game is by providing some sort of unique id which would=20
> be only meaningful to those similar game clients.
>  >
> >>>(Also if I saw the characteristics above,=20
> >>>I could as well determine that they describe normal VoIP=20
> >>>application running in (very) some old PC with half-duplex=20
> >>>audio drivers, so I claim the service description is hard to=20
> >>>make unambiguous in the first place.) =20
>=20
> I think the goal should be that all of these are at least minimally=20
> interoperable. I ought to be able to call a PTT contact from=20
> a regular=20
> voice phone and have *something* reasonable happen. (Maybe it is just=20
> oneway voice, with the non-ptt caller never getting the=20
> floor, but that=20
> may be better than nothing.
>=20
> I think the real problem in this case is not with the capability=20
> approach. It is that the 'duplex' capability is=20
> underspecified. There is=20
> no explanation of how a pair of half duplex endpoints (or one=20
> half-duplex and one full-duplex endpoint) agree on who can talk.
>=20
> In the case of the 'voice' capability, it is understood that=20
> having both=20
> endpoints support voice isn't enough to support=20
> communication. There is=20
> an SDP negotiation of codecs, etc. that either succeeds or=20
> fails. There=20
> should be something similar for half-duplex.
>=20
> >>I also have the impression that it will be difficult to=20
> determine the
> >>services merely based on the SIP methods (and ...) supported=20
> >>by the SIP UA.
> >>It possibly might mean that the SIP-based ptt-service on my=20
> >>mobile phone
> >>might be mapped onto some communication service on your PC.=20
> >>Then, I wonder
> >>who will be responsible for such kind of mapping? Only the=20
> >>end terminals, or
> >>is support from the network, presence server, necessary? And,=20
> >>how will one
> >>service in one domain be mapped into a service in another=20
> >>domain? I believe
> >>there are some open issues here.=20
>=20
> The intent is that capabilities have common meanings in all=20
> domains, so=20
> they don't need to be mapped. I don't see why there should be any=20
> binding between services and domains.
>=20
> In the general case one should assume that there are devices=20
> of widely=20
> varying capabilities in every domain, and that they know best=20
> what they=20
> are capable of. So the device should be the one deciding if a contact=20
> with and advertised set of capabilties is something it wants=20
> to attempt=20
> communication with.
>=20
> In some environment where there is an intermediary in front=20
> of a device=20
> that understands the device and acts on its behalf, then I suppose it=20
> could make these decisions. But I would consider that the odd case.
>=20
> > I think that it is fully OK to describe some rather simple=20
> things using characteristics, but in many cases applications=20
> are more complex.
>=20
> Clearly we have chosen not to describe capabilities if full detail.=20
> (E.g. we don't publish codec support.) Because of this, there is the=20
> possibility that we make a wrong decision about ability to=20
> communicate=20
> and it doesn't get discovered until we try.
>=20
> This is ok when the expected probability of failure is low. (We=20
> generally expect that there will be some codec in common, or that one=20
> end or the other will be able to introduce a transcoder.)
>=20
> >>>I think the main point of a service tuple is to give the=20
> >>>watcher enough information to know whether he can really=20
> >>>communicate & interoperate with the advertised service. Given=20
> >>>that many services taht are using SIP also have propritary or=20
> >>>non-SIP features, I don't think the current approach is enough.=20
>=20
> In cases where the existing capabilities aren't sufficient to=20
> ensure a=20
> good probability of communication, then we should indeed=20
> consider adding=20
> additional capabilities.
>=20
> But I continue to object to the notion of "service" as the way to do=20
> that, because it conveys no semantics about what the=20
> capability is that=20
> it describes.
>=20
> >>Do you have an example of a service that uses SIP and also has non
> >>SIP-features? If your service also uses non-SIP features,=20
> you can not
> >>determine the type of service by only looking at the SIP=20
> capabilities
> >>supported by the SIP UA. Some indication of that non-SIP=20
> >>feature should also
> >>be present in the presence document.=20
> >>
> >=20
> >=20
> > True, and since some of those non-SIP features might not be=20
> based on (IETF) standards, it might be good to have the=20
> possibility for the developers of such applications to have a=20
> short cut of saying what the app really is, rather than=20
> having to list characteristics. In the end, some applications=20
> are really not even meant to work with any other than the=20
> exactly same application, for instance some networking games.=20
>=20
> If the intent is to define a closed world, where=20
> interoperability is not=20
> even considered a desirable feature, then I agree. I think that may=20
> indeed be the case for proprietary games.
>=20
> But for something like PTT, while it might be in the best interest of=20
> the maker of a particular PTT solution to keep it closed, I=20
> don't think=20
> it is in the best interest of society or the IETF that this=20
> happen. This=20
> is precisely the forum in which we try to prevent that from happening.
>=20
> >>>Of course each of these proprietary services are allowed to=20
> >>>define additional status or tuple-level extensions to PIDF.
> >>
> >>Yes, you can always extend the PIDF. My comment here is that=20
> >>this might lead
> >>to an "explosion" of applications or services. It will=20
> >>certainly not help
> >>interoperability.
>=20
> My point precisely.
>=20
> > And who would standardize such characteristic definitions.=20
> It has already taken a couple of years to standardize even=20
> "prescaps", which is just a starting point.=20
> >=20
> >=20
> >>>However, I would like to have some kind of "service=20
> >>>identifier" as part of the basic framework so that this could=20
> >>>be done in a consistent manner. I think it would help=20
> >>>especially in making of authorization and composition rules=20
> >>>more simple. The current "class" attribute is way too=20
> >>loosely defined.
> >>
> >>>So the concrete proposal is to include in RPID a "service id"=20
> >>>element that would have a vendor-specific namespace, similar=20
> >>>to e.g. vendor-specific XCAP AUIDs. So for instance the=20
> >>>SIP-based PTT app from vendor xyz.com would have, e.g.=20
> >>><service-id>com.xyz.ptt</service-id>
> >>>while the PTT app compliant with OMA would have, e.g.
> >>><service-id>com.openmobilealliance.ptt</service-id>
>=20
> I am not fond of this solution. Done this way, we have defined an=20
> element that has no semantics at all, only syntax.
>=20
> I would rather see vendor specific extensions to PIDF. At=20
> least then the=20
> people that support an extension will understand what it means.
>=20
> But I would rather see people come forward and do the hard work of=20
> publicly defining new capabilities so that independent=20
> implementations=20
> can use them in an interoperable way.
>=20
> >>In general, I could agree with that approach. However, I=20
> >>don't see a reason
> >>why the vendor specific SIP-based PTT should be different=20
> >>from the OMA based
> >>PTT. Simply define one PTT, based on the SIP UA capabilities.
> >>
> >=20
> >=20
> > Well, for instance Nokia has a prorietary PTT application,=20
> which uses SIP INVITE etc., so users are essentially=20
> addressed with sip: URI. However, it has some floor control,=20
> media etc. features which make it basically non-interoperable=20
> with clients that are not specifically following that=20
> proprietary spec.=20
>=20
> This is a problem, not a good thing. I realize it has to=20
> happen as part=20
> of the development process, but the goal should be to get out=20
> of it as=20
> fast as possible.
>=20
> I would hope that endpoints supporting this would be=20
> described as well=20
> as possible using existing capabilities, and that a=20
> reasonable effort be=20
> made to do something reasonable when some other sip device connects.
>=20
> If you really can't deal with a regular voice caller at all, then I=20
> would suggest, as a transitional step, saying that you don't support=20
> voice, and then advertising some proprietary capability as an=20
> alternative to voice.
>=20
> > OMA PTT is closer to standard IETF SIP. Unless there is=20
> some conversion, it does not work with Nokia PTT directly.=20
> There should be some way for distinguishing these two in a=20
> presence document, so that there won't be confusion. I think=20
> the current idea is to do some kind of proprietary PIDF=20
> extensions, but in my opinion a standard XML element in RPID=20
> to tell this kind of information would be better, also for=20
> the purposes of authorization rule making.
>=20
> See above for what I think you could do for now. The same issues may=20
> apply to OMA PTT, and a similar solution could be applied to it.
>=20
> Hopefully somebody will come up with a PTT that can be dealt=20
> with more=20
> gracefully. I think that work should be done on better defining=20
> half-duplex, so that it can be used in conjunction with voice to=20
> describe PTT. But it may just turn out to be the wrong thing.=20
> Maybe we=20
> need a "floor-control" capability.
>=20
> >>>in _addition_ to describing sip:, half-duplex audio, and the=20
> >>>supported methods. Now:
> >>>1.) Watcher having one of those apps could see right away=20
> >>>whether it is possible to communicate
> >>>2.) Composer getting PUBLISHes from 2 sources could=20
> >>>immediately know that they are from a similar app
> >>>3.) The app could set its authorization rules using the unique id.
> >>>
> >>>The main downside I can see in this approach that if there=20
> >>>are two different proprietary apps that indeed could=20
> >>>communicate at least partially, the watcher might make a=20
> >>>conclusion that communication is not possible baed on=20
> >>>different service-id.
>=20
> This is my concern as well.
>=20
>  >>> However, in this case the watcher still
> >>>would have the characteristics visible and could determine=20
> >>>that some interworking could be achieved.=20
> >>
> >>Personally, I don't think it is a good idea that an end-user=20
> >>should make
> >>such decission. If I get a presence document that indicates=20
> >>that another
> >>person is able to receive some type of communication, I=20
> >>expect that that
> >>type of communication would actually work when I use it.
>=20
> With high probability. There are no guarantees with presence. For=20
> instance see my mention above of codecs.
>=20
> > Exactly right. If you are an application who does not=20
> necessarily aim for interoperability, but still uses sip:,=20
> INVITE etc., you should be able to explicitely say so.
>=20
> And you can, without a special notion of service.
>=20
> >>>For this kind of=20
> >>>reasons there should be careful text about when to use=20
> >>>service-id in the first place, and what can be determined from it.
> >>>
> >>>Does someone think that this is an absolutely bad idea? If=20
> >>>so, how would you envision solving the issues discussed above?
>=20
> I still don't buy this.
>=20
> >>I think more care should be taken in mapping SIP UA=20
> >>capabilities into actual
> >>services. That is not so clear to me!
> >=20
> > The idea is nice, but I don't see a problem in _addition_=20
> defining the application with something like=20
> "com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind=20
> of profile would have more meaning than the list of characteristics.
>=20
> I am still opposed to defining an attribute with no semantics.
> We have been around this so many times that I am quite=20
> certain that it=20
> will never be possible to get a useful agreement on what=20
> "service" means.
>=20
> If it is introduced, I expect it to have problems like those=20
> of the INFO=20
> method. (The arguments for introducing it seem similar.)
>=20
> 	Paul
>=20
>=20

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Hello,


At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a way to
indicate to the server which client the user uses.  I can imagine that an
operator has specific client capabilities he would like to know (e.g. java
virtual machine version, screen size).  When including this client
information, the server could retrieve the client capabilities from his e.g.
his DB.  With this extension, the operator has control over the client
information and is not dependend on the information the client provides to
him.  
Next example shows a possible extension

<ClientInfo>
	<Manufacturor> Siemens </Manufacturor>
	<Model> CX65 </Model>
	<Version>2.103564</Version>
</ClientInfo>

Regards,

Frank



------_=_NextPart_001_01C4A20A.6448CF84
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2657.73">
<TITLE>Extend UA Capabiity Extension to PIDF</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hello,</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">At this moment, =
draft-ietf-simple-prescaps-ext-01 does not provide a way to indicate to =
the server which client the user uses.&nbsp; I can imagine that an =
operator has specific client capabilities he would like to know (e.g. =
java virtual machine version, screen size).&nbsp; When including this =
client information, the server could retrieve the client capabilities =
from his e.g. his DB.&nbsp; With this extension, the operator has =
control over the client information and is not dependend on the =
information the client provides to him.&nbsp; </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Next example shows a possible =
extension</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&lt;ClientInfo&gt;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&lt;Manufacturor&gt; Siemens =
&lt;/Manufacturor&gt;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&lt;Model&gt; CX65 &lt;/Model&gt;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&lt;Version&gt;2.103564&lt;/Version&gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&lt;/ClientInfo&gt;</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Frank</FONT>
</P>
<BR>

</BODY>
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Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE documents
Date: Fri, 24 Sep 2004 11:35:46 +0300
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I'm ok with MAY. It is probably more suitable than the MUST. A consumer =
can choose not to validate the XML document and it could only hurt =
itself by not doing so.

If there are no objections or further suggestions, I'll take this text.

Regards,
Hisham

> -----Original Message-----
> From: ext Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: 23.September.2004 15:56
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: simple@ietf.org
> Subject: Re: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>=20
>=20
> Hi Hisham,
>=20
> For my money, your statements are roughly right.  The only=20
> concern I have is
> the bit where it says: "...consumer MUST validate...".  It's=20
> probably not
> the intent, but there's a potential implication that the IETF=20
> is saying that
> consumers must run a separate validation exercise on the received data
> (rather than say implicitly validating while the application=20
> interprets the
> data, in which case they may only end up validating the parts=20
> of the data
> they are interested in).  It sounds like the IETF is saying how people
> should build their devices which traditionally they haven't=20
> done.  It's
> probably not a big issue, but a possible re-wording is:-
>=20
> "The XML document received by a consumer MUST be valid according the
> schemas, including extension schemas, the consumer has access=20
> to that are
> applicable to this XML document.  The consumer MAY validate=20
> the received XML
> document against such schemas, and MAY reject the XML document if such
> validation is not successful."
>=20
> I'll leave it to you.
>=20
> Pete.
> --
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> Pete Cordell
> Tech-Know-Ware
> pete(at)tech-know-ware.com
> http://www.tech-know-ware.com
> http://www.xml2cpp.com - Now Available
> +44 1473 635863
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=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
> ----- Original Message -----=20
> From: <hisham.khartabil@nokia.com>
> To: <Pekka.Pessi@nokia.com>; <jdrosen@dynamicsoft.com>
> Cc: <Pekka.Pessi@nokia.com>; <Jari.Urpalainen@nokia.com>;=20
> <simple@ietf.org>;
> <RjS@xten.com>
> Sent: Wednesday, September 22, 2004 9:54 AM
> Subject: RE: [Simple] Strength of XML validity requirements in SIMPLE
> documents
>=20
>=20
> How does this proposal sound (for the filter drafts):
>=20
> In the format document, the following text appears: The XML=20
> document MUST be
> well-formed and MUST be valid according to schemas, including=20
> extensions
> schemas, available to the validater and applicable to the XML=20
> document.
>=20
> In the functionality draft: The creator MUST ensure that the=20
> XML document is
> well-formed and valid. The creator MUST NOT insert any=20
> extension elements or
> attributes into the XML document unless it has access to the extension
> schema and can validate the XML document. The XML document=20
> consumer MUST
> validate the XML document according the schemas, including extension
> schemas, it has access to that are applicable to this XML document.
>=20
> For pidf, the second part might be slightly different since=20
> the creator of
> the document can be the compositor and the compositor can=20
> create a presence
> XML document by aggregating tuples that contain extensions that the
> compositor is unaware of. Perhaps something can go in the=20
> Date Model draft
> discussing this.
>=20
> Regards,
> Hisham
>=20
>=20
>=20
> > -----Original Message-----
> > From: simple-bounces@ietf.org
> > [mailto:simple-bounces@ietf.org]On Behalf
> > Of ext Pekka Pessi
> > Sent: 20.September.2004 14:36
> > To: Jonathan Rosenberg
> > Cc: Pessi Pekka (Nokia-NRC/Helsinki); Urpalainen Jari
> > (Nokia-NRC/Helsinki); simple@ietf.org; Robert Sparks
> > Subject: Re: [Simple] Strength of XML validity requirements=20
> in SIMPLE
> > documents
> >
> >
> > Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:
> > >I'm saying that a document of type application/pidf+xml
> > could be valid
> > >against the PIDF schema, but it contains extensions (which
> > are allowed in
> > >the pidf schema), and those extensions make use of elements
> > defined in
> > >another schema. However, those elements are not valid
> > according to that
> > >other schema.
> >
> > >In this case, do we consider this document "valid"?
> >
> > Term valid has at least three different senses with the XML:
> >
> > 1)
> >
> > W3C defines that an XML document is valid when it has a DTD and
> > the document complies with the constraints set forth in DTD.
> >
> > That is not what we want. No <!DOCTYPE> for presence, no "MUST be
> > valid", please.
> >
> > (When RFC 3863 says that the application/pidf+xml document MUST
> > be well-formed and SHOULD be valid, I guess it just says that
> > extensions are allowed, even if they are not shown in the DTD.)
> >
> > 2)
> >
> > XML schema specification uses a bit different terminology. Schemas
> > don't specify documents, but trees (item and all its descendants).
> > You can assess validity of a particular tree, for instance, all
> > the XML stuff within <presence> element. The validator can also
> > skip certain elements (like the wildcard elements used for
> > extending pidf). So, validator can return different results: tree
> > and items within it are fully validated, partially validated, or
> > not validated at all. Also each, item within the tree are valid or
> > invalid or their validity is not known.
> >
> > I'd say that the XML document is schema-valid if its root element
> > is fully validated and it and all its descendants are valid. This
> > is not the official definition, however.
> >
> > This is unfeasible for the intermediate nodes (or processes), too.
> > Unless the node knows the schema (and semantics) of all the
> > extensions, it cannot assess the validity of composed or filtered
> > documents. I think filtering and composing is precisely the things
> > that we want to do in the server nodes.
> >
> > I can also imagine some applications that would just use existing
> > presence service, just add their non-standard application-specific
> > status in the published presence document. Now imagine two of
> > those applications running on the same node.
> >
> > 3)
> >
> > Beside validity assessment, XML schema specification has the
> > concept of "local schema-validity", too:
> >
> > <http://www.w3.org/TR/xmlschema-1/#section-Overview-of-XML-Schema>
> >
> > I must admit that I don't fully understand what this
> > means, but I have a good hunch that it is what most of us
> > want. ;)
> >
> > A element or attribute is valid if it satisfies the constraints
> > set by the XML schema. So, <presence> element is valid, if it
> > contains entity attribute, zero or more <tuple> elements, then
> > zero or more <note> elements and after then something wildcarded
> > stuff (I guess that is anything else but pidf-namespace stuff). I
> > don't understand how the datatype checking of evil types like
> > xs:ID or xs:IDREF is done when the "local schema-validity" is
> > determined. I *guess* that ID/IDREF attribute in tuple is valid if
> > its value is syntactically correct. Like id=3D"_123" is valid but,
> > id=3D"123" is not. No check for all id's within a document/item and
> > all of its descendats is required.
> >
> > So, if we have a application/pidf+xml document, we can simply say
> > that it MUST conform to its schema and other requirements set
> > forth in RFC 3863. If we want to be more verbose, we could say
> > that the root node in the document MUST be partially or fully
> > validated and none of the validated nodes MUST NOT be invalid. Or
> > we could invent our own term and say that documents MUST be
> > locally-schema-valid with the schema XXX.
> >
> > --Pekka
> >
> > _______________________________________________
> > Simple mailing list
> > Simple@ietf.org
> > https://www1.ietf.org/mailman/listinfo/simple
> >
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20
>=20

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From: Goeman Stefan <Stefan.Goeman@siemens.com>
To: "'simple@ietf.org'" <simple@ietf.org>
Subject: RE: [Simple] Presence Authorization Rules
Date: Fri, 24 Sep 2004 11:25:45 +0200
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Hello,

Comments inline.

Greetings,
Stefan.

> -----Original Message-----
> From: Tschofenig Hannes [mailto:hannes.tschofenig@siemens.com] 
> Sent: donderdag 23 september 2004 19:16
> To: Goeman Stefan; 'simple@ietf.org'
> Subject: RE: [Simple] Presence Authorization Rules
> 
> 
> # hi stefan, 
>  
> # please see my comments inline.
> 
> Hello All, 
> 
> I have been reading the draft on Presence Authorization 
> Rules, and also the
> draft draft-ietf-geopriv-common-policy. 
> I have some remarks (otherwise I would not mail of course). 
> 
> First of all, the draft on Presence Authorization Rules contracts the
> geopriv draft, at least in one place. 
> In Section 3.2 on Actions, values are given to block, 
> confirm, ... . If you
> have two rules for the same entity and you combine them 
> according to the
> combining rules as explained in the geopriv draft you end up with the 
> highest value.
> # to be more precise the combining permissions algorithms is 
> applicable if
> two or more rules fire.

Yes, I understand that, and indeed you need to do something to resolve
conflicts.
 
> 
>  Yet, in this case the highest value is the least privacy 
> preserving. This
> is in contradiction with the statement on page 21 of the 
> geopriv draft:
> 
> "Furthermore, permissions and the meaning of their values 
> MUST be defined in
> such a way that the usage of combining rules CR1, CR2, and CR3 always
> preserves or increases the level of privacy protection for the PT". 
> 
> One can argue what is wrong here. Personally, I think the 
> statement in the
> geopriv draft will not really work well (see also below).
> 
> # what would happen if jonathan changes the values 
> corresponding to the
> listed actions from
> 
> Block (0)
> Confirm (1)
> Polite-block (2)
> Allow (3)
> 
> # to the following values: 
> 
> Block (3)
> Confirm (2)
> Polite-block (1)
> Allow (0)
> 
> # now assume that one rule with allow(0) and another one with block(3)
> fires. the combining permissions algorithm would result in a block(3)
> action. 
> 
> # are you more happy with this result? 
> 
In theory, this would be OK. However, I am afraid it is a little bit
counter-intuitive. Suppose you have to groups (and groups make sense,
certainly on the terminal), friends and colleagues, and one person would be
in both groups. In this case you would have two rules for this person.
Suppose that when "not at work" you block everything for colleagues, and you
still allow some things to be seen for friends. In this case, with the
changes proposed above, would be blocked. This I find counter-intuitive. 

I still believe that it comes from combining positive (permit) rules, while
the end result will be less than the origination rules. That is what the
statement in geopriv says; and that is what I find counter-intuitive.
Or, to put it simpler, suppose you have no rules. Then nothing is visible to
other users. Then, you introduce one rule for one person, and something
becomes visible for that person. You introduce a second rule for the same
person, which results in decreasing the visibility again.  

> 
> Additionally, wouldn't it also be nice if one could group 
> people into a
> class of people and define a rule for that class of people. 
> For example, you
> could have a class of friends, family, colleagues and define 
> a rule each of
> this class. The problem know arises that one person could belong to
> different groups. Then again, the statement in the geopriv 
> draft will not
> work well. 
> # grouping people together can be solved on a user interface 
> basis. There is
> not necessarily a need to specify groups in the policies 
> itself (except if
> you want to reduce the bandwidth consumption). 
>
I don't think that is completely true. It goes further than just the user
interface. Suppose you have your buddy list, which also resides in the
network (see draft on resource lists). The buddy list is partitioned in
several groups, i.e. friends, family, ... . In your authorization policies,
you would have a rule for friends, family, ... . In this case, when a
subscribe arrives, the policy server should resolve the friends, family, and
other groups, by contacting the buddy list server, into the real entities
before you can find matching rules. I can understand that you have
objections to this practice, but it has advantages. If I add an entry in one
of my groups (friends, family, ...), I don't have to change my policy rules.
I don't see how you could solve that only based on grouping people on the
user interface; in this case, I believe you also have to change you policy
rules.


 
> ~snip~
> 
> # ciao
> # hannes
> 

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From simple-bounces@ietf.org  Fri Sep 24 05:42:53 2004
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
To: "'simple@ietf.org'" <simple@ietf.org>
Subject: RE: [Simple] Presence Authorization Rules
Date: Fri, 24 Sep 2004 11:38:52 +0200
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Hello All,

I forgot to mention something. We are talking about policies, but we are not
talking about policy enforcement (or maybe I missed that), which I believe
is also important. 
It does not make sense to block certain forms of communication from
watchers, that when they try that form of communication anyhow, that they
would succeed. This would be very annoying for the presentity.

Greetings,
Stefan. 

> -----Original Message-----
> From: Tschofenig Hannes [mailto:hannes.tschofenig@siemens.com] 
> Sent: donderdag 23 september 2004 19:16
> To: Goeman Stefan; 'simple@ietf.org'
> Subject: RE: [Simple] Presence Authorization Rules
> 
> 
> # hi stefan, 
>  
> # please see my comments inline.
> 
> Hello All, 
> 
> I have been reading the draft on Presence Authorization 
> Rules, and also the
> draft draft-ietf-geopriv-common-policy. 
> I have some remarks (otherwise I would not mail of course). 
> 
> First of all, the draft on Presence Authorization Rules contracts the
> geopriv draft, at least in one place. 
> In Section 3.2 on Actions, values are given to block, 
> confirm, ... . If you
> have two rules for the same entity and you combine them 
> according to the
> combining rules as explained in the geopriv draft you end up with the 
> highest value.
> # to be more precise the combining permissions algorithms is 
> applicable if
> two or more rules fire. 
> 
>  Yet, in this case the highest value is the least privacy 
> preserving. This
> is in contradiction with the statement on page 21 of the 
> geopriv draft:
> 
> "Furthermore, permissions and the meaning of their values 
> MUST be defined in
> such a way that the usage of combining rules CR1, CR2, and CR3 always
> preserves or increases the level of privacy protection for the PT". 
> 
> One can argue what is wrong here. Personally, I think the 
> statement in the
> geopriv draft will not really work well (see also below).
> 
> # what would happen if jonathan changes the values 
> corresponding to the
> listed actions from
> 
> Block (0)
> Confirm (1)
> Polite-block (2)
> Allow (3)
> 
> # to the following values: 
> 
> Block (3)
> Confirm (2)
> Polite-block (1)
> Allow (0)
> 
> # now assume that one rule with allow(0) and another one with block(3)
> fires. the combining permissions algorithm would result in a block(3)
> action. 
> 
> # are you more happy with this result? 
> 
> 
> Additionally, wouldn't it also be nice if one could group 
> people into a
> class of people and define a rule for that class of people. 
> For example, you
> could have a class of friends, family, colleagues and define 
> a rule each of
> this class. The problem know arises that one person could belong to
> different groups. Then again, the statement in the geopriv 
> draft will not
> work well. 
> # grouping people together can be solved on a user interface 
> basis. There is
> not necessarily a need to specify groups in the policies 
> itself (except if
> you want to reduce the bandwidth consumption). 
> 
> ~snip~
> 
> # ciao
> # hannes
> 

_______________________________________________
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Simple@ietf.org
https://www1.ietf.org/mailman/listinfo/simple


From simple-bounces@ietf.org  Fri Sep 24 07:46:43 2004
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
To: "'simple@ietf.org'" <simple@ietf.org>
Subject: RE: [Simple] Presence Data Model: Identifying services
Date: Fri, 24 Sep 2004 13:40:10 +0200
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Hello All,

I would like to add the following questions/problems to the discussion.

How will a presence document be presented to an end-user. Personally, I
don't think it makes much sense to present a "raw" presence document, like
the one on page 26 of the Presence Data Model draft, to the end-user. In the
end, people would like to see on their device: "I can contact person X via
email, voice-call, IM, ...". I don't think they would like to see things
like: "I can contact person X via SIP". I believe that 99.99% of the world
does not know what SIP is, yet everybody (well ...) knows what email is. 

The point I want to make is that some entity has to process a presence
document in a way that it can be presented to an end-user in an meaningful
manner. 

My first guess was that a device could the transformation of the "raw"
presence document. Device know which applications they are currently
running. So, there should be no problem here.

On second thought, I realized that there is one problem:
How will this work together with the Authorization rules. Here, again, you
have to present the authorization rules in a meaningful manner (via some
front end user interface). If I am correct, the idea is to retrieve the
authorization rules via XCAP from the policy server, make some changes and
provide the changes back to the policy server via XCAP.  Yet, the policy
server is not aware of any services like IM, voice call, ... . So, again,
the device should make this mapping. This works well if you only use one
device. But, if you use more than one device, with different capabilities
and services, it does not work anymore. A device can't say something
meaningful over a service it does not support. Consequently, you can not
edit every aspect of the authorization rules on your device. I think this is
problematic, and I don't see a good solution right know.

Greetings,
Stefan. 

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: donderdag 23 september 2004 17:47
> To: Markus.Isomaki@nokia.com
> Cc: Goeman Stefan; simple@ietf.org
> Subject: Re: [Simple] Presence Data Model: Identifying services
> 
> 
> I have comments both to Markus' original posting and to the ensuing 
> discussion. So I will just insert inline in the latest 
> response I have.
> 
> 	Paul
> 
> Markus.Isomaki@nokia.com wrote:
> > Hi Stefan,
> > 
> > Comments back to you inline. 
> > 
> > (Yes, I absolutely agree with you on the usefulness of the 
> Presence Data Model draft and that it should have been the 
> starting point, at least right after PIDF.)
> > 
> > 
> >>-----Original Message-----
> >>From: simple-bounces@ietf.org 
> >>[mailto:simple-bounces@ietf.org]On Behalf
> >>Of ext Goeman Stefan
> >>Sent: 23 September, 2004 14:42
> >>To: 'simple@ietf.org'
> >>Subject: RE: [Simple] Presence Data Model: Identifying services
> >>
> >>
> >>Hello,
> >>
> >>Comments inline
> >>
> >>P.S.: In general, I find the Presence Data model draft a good 
> >>document. This
> >>is how things should have started in the first place.
> >>
> >>Greetings,
> >>Stefan.
> >>
> >>
> >>>-----Original Message-----
> >>>From: simple-bounces@ietf.org 
> >>>[mailto:simple-bounces@ietf.org] On Behalf Of 
> >>
> >>Markus.Isomaki@nokia.com
> >>
> >>>Sent: donderdag 23 september 2004 10:10
> >>>To: simple@ietf.org
> >>>Subject: [Simple] Presence Data Model: Identifying services
> >>>
> >>>
> >>>Hi all,
> >>>
> >>>I have a comment on Presence Data Model draft about the 
> >>>identification/characterization of services. (I know this has 
> >>>been discussed for a while already, but I still think the 
> >>>current way is somewhat unadequate.)
> >>>
> >>>The current Data Model draft says:
> >>>
> >>>   In this model, services are not explicitly enumerated.  That is,
> >>>   there is no "service" attribute with values of "ptt" or 
> >>>"telephony".
> >>>   Rather, the service is identified in one of two ways.  In 
> >>>many cases,
> >>>   the URI scheme is a clear indicator of service.  An "sms" 
> >>>URI clearly
> >>>   indicates SMS, and a "tel" URI clearly indicates 
> >>>telephony. 
> 
> I don't find the scheme as clearcut as that. I don't think one should 
> assume "tel" indicates telephony. It perhaps does imply 
> reachability via 
> various telephony protocols for purpose of voice 
> communications, but is 
> not exclusive to that. If the tel: URI can be mapped to a sip 
> endpoint 
> and protocol, then non-voice, non-telephony capabilities are 
> not excluded.
> 
> For instance, it may prove convenient to publish a tel URI for a sip 
> endpoint that is capable of both voice and IM. Then calls via 
> the PSTN 
> will work, calls from a sip audio phone will work, calls from another 
> voice+im client will have access to voice and IM.
> 
>  >>> For some
> >>>   URIs, there may be many services available, for example, 
> >>
> >>SIP.  For
> >>
> >>>   those services, each service has a set of 
> >>
> >>characteristics, each of
> >>
> >>>   which has a well-defined meaning, such that a system can
> >>>   unequivocally determine whether or not the service has that
> >>>   characteristic.  This is discussed in more detail in [5]. 
> >>>
> >>>I agree that the contact URI scheme is the most important 
> >>>piece of information to distinguish what is the "service". 
> >>>(There are some gaps here too, since e.g. the URI schemes for 
> >>>SMS or MMS do not even exist, and there may be other non-IETF 
> >>>protocols that face the same problem.) However, I'm not 
> >>>convinced that listing the signaling/media characteristics of 
> >>>the end-point or service really gives enough information to 
> >>>the watcher to really determine if it can communicate with 
> >>>the advertised service/application. 
> >>>
> >>>The refenced draft [5] has a following example:
> >>>
> >>>5.6 Walkie-talkie
> >>>
> >>>   The walkie-talkie service allows real-time voice communication
> >>>   between participants.  Only one participant can speak at a 
> >>>time; that
> >>>   is, communication is half-duplex.  Typically, 
> >>
> >>participants press a
> >>
> >>>   button to indicate that they are ready to speak, although other
> >>>   mechanism (e.g.  voice activation) are occasionally used.
> >>>
> >>>   Support for the Walkie-Talkie service can be deduced by 
> >>>observing the
> >>>   presence of a contact URI with a scheme of "sip:", 
> >>
> >>associated with
> >>
> >>>   the following minimal set of capabilities: <audio>true</audio>
> >>>   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> >>>
> >>>Presumably this is the same as the service sometimes called 
> >>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
> >>>are several _non-interworking_ variants of this service, that 
> >>>still all would match the characteristics listed above, as 
> >>>the _SIP_ part might still be standard, at least for some 
> >>>methods etc. But still the communication would fail even if 
> >>>the tuple looks OK.
> >>
> >>I don't think I understand you here. I would expect that if 
> >>both end-point
> >>SIP US support exactly the same methods, the SIP UA should 
> >>also be able to
> >>handle those SIP method. Or, to phase it in another way, both 
> >>end-point
> >>should be able to communicate with each other only based on 
> >>using these
> >>capabilities.
> >>
> > 
> > 
> > To some extent that might be possible, but there might be 
> differences e.g. in floor control etc., which make the 
> service still practically unusable, which means that the 
> presence document shouldn't give the impression that the 
> communication will work. Obviously it is possible to list all 
> those additional characteristics as well (and this might be a 
> good idea too), but for instance if presence of a network 
> game is published, it is easier to _explicitely_ tell what 
> the game is by providing some sort of unique id which would 
> be only meaningful to those similar game clients.
>  >
> >>>(Also if I saw the characteristics above, 
> >>>I could as well determine that they describe normal VoIP 
> >>>application running in (very) some old PC with half-duplex 
> >>>audio drivers, so I claim the service description is hard to 
> >>>make unambiguous in the first place.)  
> 
> I think the goal should be that all of these are at least minimally 
> interoperable. I ought to be able to call a PTT contact from 
> a regular 
> voice phone and have *something* reasonable happen. (Maybe it is just 
> oneway voice, with the non-ptt caller never getting the 
> floor, but that 
> may be better than nothing.
> 
> I think the real problem in this case is not with the capability 
> approach. It is that the 'duplex' capability is 
> underspecified. There is 
> no explanation of how a pair of half duplex endpoints (or one 
> half-duplex and one full-duplex endpoint) agree on who can talk.
> 
> In the case of the 'voice' capability, it is understood that 
> having both 
> endpoints support voice isn't enough to support 
> communication. There is 
> an SDP negotiation of codecs, etc. that either succeeds or 
> fails. There 
> should be something similar for half-duplex.
> 
> >>I also have the impression that it will be difficult to 
> determine the
> >>services merely based on the SIP methods (and ...) supported 
> >>by the SIP UA.
> >>It possibly might mean that the SIP-based ptt-service on my 
> >>mobile phone
> >>might be mapped onto some communication service on your PC. 
> >>Then, I wonder
> >>who will be responsible for such kind of mapping? Only the 
> >>end terminals, or
> >>is support from the network, presence server, necessary? And, 
> >>how will one
> >>service in one domain be mapped into a service in another 
> >>domain? I believe
> >>there are some open issues here. 
> 
> The intent is that capabilities have common meanings in all 
> domains, so 
> they don't need to be mapped. I don't see why there should be any 
> binding between services and domains.
> 
> In the general case one should assume that there are devices 
> of widely 
> varying capabilities in every domain, and that they know best 
> what they 
> are capable of. So the device should be the one deciding if a contact 
> with and advertised set of capabilties is something it wants 
> to attempt 
> communication with.
> 
> In some environment where there is an intermediary in front 
> of a device 
> that understands the device and acts on its behalf, then I suppose it 
> could make these decisions. But I would consider that the odd case.
> 
> > I think that it is fully OK to describe some rather simple 
> things using characteristics, but in many cases applications 
> are more complex.
> 
> Clearly we have chosen not to describe capabilities if full detail. 
> (E.g. we don't publish codec support.) Because of this, there is the 
> possibility that we make a wrong decision about ability to 
> communicate 
> and it doesn't get discovered until we try.
> 
> This is ok when the expected probability of failure is low. (We 
> generally expect that there will be some codec in common, or that one 
> end or the other will be able to introduce a transcoder.)
> 
> >>>I think the main point of a service tuple is to give the 
> >>>watcher enough information to know whether he can really 
> >>>communicate & interoperate with the advertised service. Given 
> >>>that many services taht are using SIP also have propritary or 
> >>>non-SIP features, I don't think the current approach is enough. 
> 
> In cases where the existing capabilities aren't sufficient to 
> ensure a 
> good probability of communication, then we should indeed 
> consider adding 
> additional capabilities.
> 
> But I continue to object to the notion of "service" as the way to do 
> that, because it conveys no semantics about what the 
> capability is that 
> it describes.
> 
> >>Do you have an example of a service that uses SIP and also has non
> >>SIP-features? If your service also uses non-SIP features, 
> you can not
> >>determine the type of service by only looking at the SIP 
> capabilities
> >>supported by the SIP UA. Some indication of that non-SIP 
> >>feature should also
> >>be present in the presence document. 
> >>
> > 
> > 
> > True, and since some of those non-SIP features might not be 
> based on (IETF) standards, it might be good to have the 
> possibility for the developers of such applications to have a 
> short cut of saying what the app really is, rather than 
> having to list characteristics. In the end, some applications 
> are really not even meant to work with any other than the 
> exactly same application, for instance some networking games. 
> 
> If the intent is to define a closed world, where 
> interoperability is not 
> even considered a desirable feature, then I agree. I think that may 
> indeed be the case for proprietary games.
> 
> But for something like PTT, while it might be in the best interest of 
> the maker of a particular PTT solution to keep it closed, I 
> don't think 
> it is in the best interest of society or the IETF that this 
> happen. This 
> is precisely the forum in which we try to prevent that from happening.
> 
> >>>Of course each of these proprietary services are allowed to 
> >>>define additional status or tuple-level extensions to PIDF.
> >>
> >>Yes, you can always extend the PIDF. My comment here is that 
> >>this might lead
> >>to an "explosion" of applications or services. It will 
> >>certainly not help
> >>interoperability.
> 
> My point precisely.
> 
> > And who would standardize such characteristic definitions. 
> It has already taken a couple of years to standardize even 
> "prescaps", which is just a starting point. 
> > 
> > 
> >>>However, I would like to have some kind of "service 
> >>>identifier" as part of the basic framework so that this could 
> >>>be done in a consistent manner. I think it would help 
> >>>especially in making of authorization and composition rules 
> >>>more simple. The current "class" attribute is way too 
> >>loosely defined.
> >>
> >>>So the concrete proposal is to include in RPID a "service id" 
> >>>element that would have a vendor-specific namespace, similar 
> >>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
> >>>SIP-based PTT app from vendor xyz.com would have, e.g. 
> >>><service-id>com.xyz.ptt</service-id>
> >>>while the PTT app compliant with OMA would have, e.g.
> >>><service-id>com.openmobilealliance.ptt</service-id>
> 
> I am not fond of this solution. Done this way, we have defined an 
> element that has no semantics at all, only syntax.
> 
> I would rather see vendor specific extensions to PIDF. At 
> least then the 
> people that support an extension will understand what it means.
> 
> But I would rather see people come forward and do the hard work of 
> publicly defining new capabilities so that independent 
> implementations 
> can use them in an interoperable way.
> 
> >>In general, I could agree with that approach. However, I 
> >>don't see a reason
> >>why the vendor specific SIP-based PTT should be different 
> >>from the OMA based
> >>PTT. Simply define one PTT, based on the SIP UA capabilities.
> >>
> > 
> > 
> > Well, for instance Nokia has a prorietary PTT application, 
> which uses SIP INVITE etc., so users are essentially 
> addressed with sip: URI. However, it has some floor control, 
> media etc. features which make it basically non-interoperable 
> with clients that are not specifically following that 
> proprietary spec. 
> 
> This is a problem, not a good thing. I realize it has to 
> happen as part 
> of the development process, but the goal should be to get out 
> of it as 
> fast as possible.
> 
> I would hope that endpoints supporting this would be 
> described as well 
> as possible using existing capabilities, and that a 
> reasonable effort be 
> made to do something reasonable when some other sip device connects.
> 
> If you really can't deal with a regular voice caller at all, then I 
> would suggest, as a transitional step, saying that you don't support 
> voice, and then advertising some proprietary capability as an 
> alternative to voice.
> 
> > OMA PTT is closer to standard IETF SIP. Unless there is 
> some conversion, it does not work with Nokia PTT directly. 
> There should be some way for distinguishing these two in a 
> presence document, so that there won't be confusion. I think 
> the current idea is to do some kind of proprietary PIDF 
> extensions, but in my opinion a standard XML element in RPID 
> to tell this kind of information would be better, also for 
> the purposes of authorization rule making.
> 
> See above for what I think you could do for now. The same issues may 
> apply to OMA PTT, and a similar solution could be applied to it.
> 
> Hopefully somebody will come up with a PTT that can be dealt 
> with more 
> gracefully. I think that work should be done on better defining 
> half-duplex, so that it can be used in conjunction with voice to 
> describe PTT. But it may just turn out to be the wrong thing. 
> Maybe we 
> need a "floor-control" capability.
> 
> >>>in _addition_ to describing sip:, half-duplex audio, and the 
> >>>supported methods. Now:
> >>>1.) Watcher having one of those apps could see right away 
> >>>whether it is possible to communicate
> >>>2.) Composer getting PUBLISHes from 2 sources could 
> >>>immediately know that they are from a similar app
> >>>3.) The app could set its authorization rules using the unique id.
> >>>
> >>>The main downside I can see in this approach that if there 
> >>>are two different proprietary apps that indeed could 
> >>>communicate at least partially, the watcher might make a 
> >>>conclusion that communication is not possible baed on 
> >>>different service-id.
> 
> This is my concern as well.
> 
>  >>> However, in this case the watcher still
> >>>would have the characteristics visible and could determine 
> >>>that some interworking could be achieved. 
> >>
> >>Personally, I don't think it is a good idea that an end-user 
> >>should make
> >>such decission. If I get a presence document that indicates 
> >>that another
> >>person is able to receive some type of communication, I 
> >>expect that that
> >>type of communication would actually work when I use it.
> 
> With high probability. There are no guarantees with presence. For 
> instance see my mention above of codecs.
> 
> > Exactly right. If you are an application who does not 
> necessarily aim for interoperability, but still uses sip:, 
> INVITE etc., you should be able to explicitely say so.
> 
> And you can, without a special notion of service.
> 
> >>>For this kind of 
> >>>reasons there should be careful text about when to use 
> >>>service-id in the first place, and what can be determined from it.
> >>>
> >>>Does someone think that this is an absolutely bad idea? If 
> >>>so, how would you envision solving the issues discussed above?
> 
> I still don't buy this.
> 
> >>I think more care should be taken in mapping SIP UA 
> >>capabilities into actual
> >>services. That is not so clear to me!
> > 
> > The idea is nice, but I don't see a problem in _addition_ 
> defining the application with something like 
> "com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind 
> of profile would have more meaning than the list of characteristics.
> 
> I am still opposed to defining an attribute with no semantics.
> We have been around this so many times that I am quite 
> certain that it 
> will never be possible to get a useful agreement on what 
> "service" means.
> 
> If it is introduced, I expect it to have problems like those 
> of the INFO 
> method. (The arguments for introducing it seem similar.)
> 
> 	Paul
> 

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Subject: Re: [Simple] Presence Data Model: Identifying services
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Paul Kyzivat <pkyzivat@cisco.com>
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On Thu, 2004-09-23 at 18:46, ext Paul Kyzivat wrote:
> I don't find the scheme as clearcut as that. I don't think one should 
> assume "tel" indicates telephony. It perhaps does imply reachability via 
> various telephony protocols for purpose of voice communications, but is 
> not exclusive to that. If the tel: URI can be mapped to a sip endpoint 
> and protocol, then non-voice, non-telephony capabilities are not excluded.

Hmm... If the scheme is that ambiguous, then perhaps its definition
needs to improve.

> For instance, it may prove convenient to publish a tel URI for a sip 
> endpoint that is capable of both voice and IM. Then calls via the PSTN 
> will work, calls from a sip audio phone will work, calls from another 
> voice+im client will have access to voice and IM.

IMO the most obvious solution here is that if the endpoint wants to
advertize services other than (or in addition to) telephony, it
publishes a SIP or IM tuple as well as a tel tuple. Is there a reason
this would not work?

Cheers,
Aki

>  >>> For some
> >>>   URIs, there may be many services available, for example, 
> >>
> >>SIP.  For
> >>
> >>>   those services, each service has a set of 
> >>
> >>characteristics, each of
> >>
> >>>   which has a well-defined meaning, such that a system can
> >>>   unequivocally determine whether or not the service has that
> >>>   characteristic.  This is discussed in more detail in [5]. 
> >>>
> >>>I agree that the contact URI scheme is the most important 
> >>>piece of information to distinguish what is the "service". 
> >>>(There are some gaps here too, since e.g. the URI schemes for 
> >>>SMS or MMS do not even exist, and there may be other non-IETF 
> >>>protocols that face the same problem.) However, I'm not 
> >>>convinced that listing the signaling/media characteristics of 
> >>>the end-point or service really gives enough information to 
> >>>the watcher to really determine if it can communicate with 
> >>>the advertised service/application. 
> >>>
> >>>The refenced draft [5] has a following example:
> >>>
> >>>5.6 Walkie-talkie
> >>>
> >>>   The walkie-talkie service allows real-time voice communication
> >>>   between participants.  Only one participant can speak at a 
> >>>time; that
> >>>   is, communication is half-duplex.  Typically, 
> >>
> >>participants press a
> >>
> >>>   button to indicate that they are ready to speak, although other
> >>>   mechanism (e.g.  voice activation) are occasionally used.
> >>>
> >>>   Support for the Walkie-Talkie service can be deduced by 
> >>>observing the
> >>>   presence of a contact URI with a scheme of "sip:", 
> >>
> >>associated with
> >>
> >>>   the following minimal set of capabilities: <audio>true</audio>
> >>>   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> >>>
> >>>Presumably this is the same as the service sometimes called 
> >>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
> >>>are several _non-interworking_ variants of this service, that 
> >>>still all would match the characteristics listed above, as 
> >>>the _SIP_ part might still be standard, at least for some 
> >>>methods etc. But still the communication would fail even if 
> >>>the tuple looks OK.
> >>
> >>I don't think I understand you here. I would expect that if 
> >>both end-point
> >>SIP US support exactly the same methods, the SIP UA should 
> >>also be able to
> >>handle those SIP method. Or, to phase it in another way, both 
> >>end-point
> >>should be able to communicate with each other only based on 
> >>using these
> >>capabilities.
> >>
> > 
> > 
> > To some extent that might be possible, but there might be differences e.g. in floor control etc., which make the service still practically unusable, which means that the presence document shouldn't give the impression that the communication will work. Obviously it is possible to list all those additional characteristics as well (and this might be a good idea too), but for instance if presence of a network game is published, it is easier to _explicitely_ tell what the game is by providing some sort of unique id which would be only meaningful to those similar game clients.
>  >
> >>>(Also if I saw the characteristics above, 
> >>>I could as well determine that they describe normal VoIP 
> >>>application running in (very) some old PC with half-duplex 
> >>>audio drivers, so I claim the service description is hard to 
> >>>make unambiguous in the first place.)  
> 
> I think the goal should be that all of these are at least minimally 
> interoperable. I ought to be able to call a PTT contact from a regular 
> voice phone and have *something* reasonable happen. (Maybe it is just 
> oneway voice, with the non-ptt caller never getting the floor, but that 
> may be better than nothing.
> 
> I think the real problem in this case is not with the capability 
> approach. It is that the 'duplex' capability is underspecified. There is 
> no explanation of how a pair of half duplex endpoints (or one 
> half-duplex and one full-duplex endpoint) agree on who can talk.
> 
> In the case of the 'voice' capability, it is understood that having both 
> endpoints support voice isn't enough to support communication. There is 
> an SDP negotiation of codecs, etc. that either succeeds or fails. There 
> should be something similar for half-duplex.
> 
> >>I also have the impression that it will be difficult to determine the
> >>services merely based on the SIP methods (and ...) supported 
> >>by the SIP UA.
> >>It possibly might mean that the SIP-based ptt-service on my 
> >>mobile phone
> >>might be mapped onto some communication service on your PC. 
> >>Then, I wonder
> >>who will be responsible for such kind of mapping? Only the 
> >>end terminals, or
> >>is support from the network, presence server, necessary? And, 
> >>how will one
> >>service in one domain be mapped into a service in another 
> >>domain? I believe
> >>there are some open issues here. 
> 
> The intent is that capabilities have common meanings in all domains, so 
> they don't need to be mapped. I don't see why there should be any 
> binding between services and domains.
> 
> In the general case one should assume that there are devices of widely 
> varying capabilities in every domain, and that they know best what they 
> are capable of. So the device should be the one deciding if a contact 
> with and advertised set of capabilties is something it wants to attempt 
> communication with.
> 
> In some environment where there is an intermediary in front of a device 
> that understands the device and acts on its behalf, then I suppose it 
> could make these decisions. But I would consider that the odd case.
> 
> > I think that it is fully OK to describe some rather simple things using characteristics, but in many cases applications are more complex.
> 
> Clearly we have chosen not to describe capabilities if full detail. 
> (E.g. we don't publish codec support.) Because of this, there is the 
> possibility that we make a wrong decision about ability to communicate 
> and it doesn't get discovered until we try.
> 
> This is ok when the expected probability of failure is low. (We 
> generally expect that there will be some codec in common, or that one 
> end or the other will be able to introduce a transcoder.)
> 
> >>>I think the main point of a service tuple is to give the 
> >>>watcher enough information to know whether he can really 
> >>>communicate & interoperate with the advertised service. Given 
> >>>that many services taht are using SIP also have propritary or 
> >>>non-SIP features, I don't think the current approach is enough. 
> 
> In cases where the existing capabilities aren't sufficient to ensure a 
> good probability of communication, then we should indeed consider adding 
> additional capabilities.
> 
> But I continue to object to the notion of "service" as the way to do 
> that, because it conveys no semantics about what the capability is that 
> it describes.
> 
> >>Do you have an example of a service that uses SIP and also has non
> >>SIP-features? If your service also uses non-SIP features, you can not
> >>determine the type of service by only looking at the SIP capabilities
> >>supported by the SIP UA. Some indication of that non-SIP 
> >>feature should also
> >>be present in the presence document. 
> >>
> > 
> > 
> > True, and since some of those non-SIP features might not be based on (IETF) standards, it might be good to have the possibility for the developers of such applications to have a short cut of saying what the app really is, rather than having to list characteristics. In the end, some applications are really not even meant to work with any other than the exactly same application, for instance some networking games. 
> 
> If the intent is to define a closed world, where interoperability is not 
> even considered a desirable feature, then I agree. I think that may 
> indeed be the case for proprietary games.
> 
> But for something like PTT, while it might be in the best interest of 
> the maker of a particular PTT solution to keep it closed, I don't think 
> it is in the best interest of society or the IETF that this happen. This 
> is precisely the forum in which we try to prevent that from happening.
> 
> >>>Of course each of these proprietary services are allowed to 
> >>>define additional status or tuple-level extensions to PIDF.
> >>
> >>Yes, you can always extend the PIDF. My comment here is that 
> >>this might lead
> >>to an "explosion" of applications or services. It will 
> >>certainly not help
> >>interoperability.
> 
> My point precisely.
> 
> > And who would standardize such characteristic definitions. It has already taken a couple of years to standardize even "prescaps", which is just a starting point. 
> > 
> > 
> >>>However, I would like to have some kind of "service 
> >>>identifier" as part of the basic framework so that this could 
> >>>be done in a consistent manner. I think it would help 
> >>>especially in making of authorization and composition rules 
> >>>more simple. The current "class" attribute is way too 
> >>loosely defined.
> >>
> >>>So the concrete proposal is to include in RPID a "service id" 
> >>>element that would have a vendor-specific namespace, similar 
> >>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
> >>>SIP-based PTT app from vendor xyz.com would have, e.g. 
> >>><service-id>com.xyz.ptt</service-id>
> >>>while the PTT app compliant with OMA would have, e.g.
> >>><service-id>com.openmobilealliance.ptt</service-id>
> 
> I am not fond of this solution. Done this way, we have defined an 
> element that has no semantics at all, only syntax.
> 
> I would rather see vendor specific extensions to PIDF. At least then the 
> people that support an extension will understand what it means.
> 
> But I would rather see people come forward and do the hard work of 
> publicly defining new capabilities so that independent implementations 
> can use them in an interoperable way.
> 
> >>In general, I could agree with that approach. However, I 
> >>don't see a reason
> >>why the vendor specific SIP-based PTT should be different 
> >>from the OMA based
> >>PTT. Simply define one PTT, based on the SIP UA capabilities.
> >>
> > 
> > 
> > Well, for instance Nokia has a prorietary PTT application, which uses SIP INVITE etc., so users are essentially addressed with sip: URI. However, it has some floor control, media etc. features which make it basically non-interoperable with clients that are not specifically following that proprietary spec. 
> 
> This is a problem, not a good thing. I realize it has to happen as part 
> of the development process, but the goal should be to get out of it as 
> fast as possible.
> 
> I would hope that endpoints supporting this would be described as well 
> as possible using existing capabilities, and that a reasonable effort be 
> made to do something reasonable when some other sip device connects.
> 
> If you really can't deal with a regular voice caller at all, then I 
> would suggest, as a transitional step, saying that you don't support 
> voice, and then advertising some proprietary capability as an 
> alternative to voice.
> 
> > OMA PTT is closer to standard IETF SIP. Unless there is some conversion, it does not work with Nokia PTT directly. There should be some way for distinguishing these two in a presence document, so that there won't be confusion. I think the current idea is to do some kind of proprietary PIDF extensions, but in my opinion a standard XML element in RPID to tell this kind of information would be better, also for the purposes of authorization rule making.
> 
> See above for what I think you could do for now. The same issues may 
> apply to OMA PTT, and a similar solution could be applied to it.
> 
> Hopefully somebody will come up with a PTT that can be dealt with more 
> gracefully. I think that work should be done on better defining 
> half-duplex, so that it can be used in conjunction with voice to 
> describe PTT. But it may just turn out to be the wrong thing. Maybe we 
> need a "floor-control" capability.
> 
> >>>in _addition_ to describing sip:, half-duplex audio, and the 
> >>>supported methods. Now:
> >>>1.) Watcher having one of those apps could see right away 
> >>>whether it is possible to communicate
> >>>2.) Composer getting PUBLISHes from 2 sources could 
> >>>immediately know that they are from a similar app
> >>>3.) The app could set its authorization rules using the unique id.
> >>>
> >>>The main downside I can see in this approach that if there 
> >>>are two different proprietary apps that indeed could 
> >>>communicate at least partially, the watcher might make a 
> >>>conclusion that communication is not possible baed on 
> >>>different service-id.
> 
> This is my concern as well.
> 
>  >>> However, in this case the watcher still
> >>>would have the characteristics visible and could determine 
> >>>that some interworking could be achieved. 
> >>
> >>Personally, I don't think it is a good idea that an end-user 
> >>should make
> >>such decission. If I get a presence document that indicates 
> >>that another
> >>person is able to receive some type of communication, I 
> >>expect that that
> >>type of communication would actually work when I use it.
> 
> With high probability. There are no guarantees with presence. For 
> instance see my mention above of codecs.
> 
> > Exactly right. If you are an application who does not necessarily aim for interoperability, but still uses sip:, INVITE etc., you should be able to explicitely say so.
> 
> And you can, without a special notion of service.
> 
> >>>For this kind of 
> >>>reasons there should be careful text about when to use 
> >>>service-id in the first place, and what can be determined from it.
> >>>
> >>>Does someone think that this is an absolutely bad idea? If 
> >>>so, how would you envision solving the issues discussed above?
> 
> I still don't buy this.
> 
> >>I think more care should be taken in mapping SIP UA 
> >>capabilities into actual
> >>services. That is not so clear to me!
> > 
> > The idea is nice, but I don't see a problem in _addition_ defining the application with something like "com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind of profile would have more meaning than the list of characteristics.
> 
> I am still opposed to defining an attribute with no semantics.
> We have been around this so many times that I am quite certain that it 
> will never be possible to get a useful agreement on what "service" means.
> 
> If it is introduced, I expect it to have problems like those of the INFO 
> method. (The arguments for introducing it seem similar.)
> 
> 	Paul
> 
> 
> _______________________________________________
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Subject: Re: [Simple] Extend UA Capabiity Extension to PIDF
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Helsen Frank <Frank.Helsen@siemens.com>
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Hi,

The primary consumer of presence documents is a presence watcher. And it
seems this information wouldn't be all that useful to most if not all
watchers.

What is the actual scenario where the server would need this
information? Because in many cases, the User-Agent header field in SIP
is supposed to carry this sort of information already.

Cheers,
Aki

On Fri, 2004-09-24 at 10:44, ext Helsen Frank wrote:
> Hello,
> 
> 
> At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a
> way to indicate to the server which client the user uses.  I can
> imagine that an operator has specific client capabilities he would
> like to know (e.g. java virtual machine version, screen size).  When
> including this client information, the server could retrieve the
> client capabilities from his e.g. his DB.  With this extension, the
> operator has control over the client information and is not dependend
> on the information the client provides to him.  
> 
> Next example shows a possible extension
> 
> <ClientInfo>
>         <Manufacturor> Siemens </Manufacturor>
>         <Model> CX65 </Model>
>         <Version>2.103564</Version>
> </ClientInfo>
> 
> Regards,
> 
> Frank
> 
> 
> 
> 
> ______________________________________________________________________
> _______________________________________________
> Simple mailing list
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Subject: Re: [Simple] Presence Data Model: Identifying services
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Aki Niemi wrote:
> On Thu, 2004-09-23 at 18:46, ext Paul Kyzivat wrote:
> 
>>I don't find the scheme as clearcut as that. I don't think one should 
>>assume "tel" indicates telephony. It perhaps does imply reachability via 
>>various telephony protocols for purpose of voice communications, but is 
>>not exclusive to that. If the tel: URI can be mapped to a sip endpoint 
>>and protocol, then non-voice, non-telephony capabilities are not excluded.
> 
> Hmm... If the scheme is that ambiguous, then perhaps its definition
> needs to improve.

Why? There is no particular reason why tel: should mean voice. We don't 
restrict sip: that way, and http: doesn't say much about what you can do 
either. (I can use http to exchange html, vxml, xcap, etc.)

AFAIK, tel: is an addressing format and namespace. The following is from 
  draft-ietf-iptel-rfc2806bis-09:

    The "tel" URI telephone number is not restricted in the type of
    termination point it refers to.  The termination point can be in the
    public telephone network, a private telephone network or the
    Internet. The termination point can be fixed or wireless and address
    a fixed wired, mobile or nomadic terminal.  The terminal addressed
    can support any electronic communication service (ECS) including
    voice, data and fax.

>>For instance, it may prove convenient to publish a tel URI for a sip 
>>endpoint that is capable of both voice and IM. Then calls via the PSTN 
>>will work, calls from a sip audio phone will work, calls from another 
>>voice+im client will have access to voice and IM.
> 
> IMO the most obvious solution here is that if the endpoint wants to
> advertize services other than (or in addition to) telephony, it
> publishes a SIP or IM tuple as well as a tel tuple. Is there a reason
> this would not work?

I don't see a problem here, and so I don't feel a need for a solution.

I agree that other address forms are probably desirable in this case, 
for a variety of reasons. But that doesn't mean we should automatically 
assume tel: implies voice telephony and only that.

One place where this situation may well come up is when the tel: uri is 
used as the presentity address. (Something that should be ok, and makes 
sense to use if you are a client and that is all you have.) It may be 
that the user doesn't want to expose device addresses, and so provides 
only the presentity address back as the contact in the tuple(s).

	Paul

> Cheers,
> Aki
> 
> 
>> >>> For some
>>
>>>>>  URIs, there may be many services available, for example, 
>>>>
>>>>SIP.  For
>>>>
>>>>
>>>>>  those services, each service has a set of 
>>>>
>>>>characteristics, each of
>>>>
>>>>
>>>>>  which has a well-defined meaning, such that a system can
>>>>>  unequivocally determine whether or not the service has that
>>>>>  characteristic.  This is discussed in more detail in [5]. 
>>>>>
>>>>>I agree that the contact URI scheme is the most important 
>>>>>piece of information to distinguish what is the "service". 
>>>>>(There are some gaps here too, since e.g. the URI schemes for 
>>>>>SMS or MMS do not even exist, and there may be other non-IETF 
>>>>>protocols that face the same problem.) However, I'm not 
>>>>>convinced that listing the signaling/media characteristics of 
>>>>>the end-point or service really gives enough information to 
>>>>>the watcher to really determine if it can communicate with 
>>>>>the advertised service/application. 
>>>>>
>>>>>The refenced draft [5] has a following example:
>>>>>
>>>>>5.6 Walkie-talkie
>>>>>
>>>>>  The walkie-talkie service allows real-time voice communication
>>>>>  between participants.  Only one participant can speak at a 
>>>>>time; that
>>>>>  is, communication is half-duplex.  Typically, 
>>>>
>>>>participants press a
>>>>
>>>>
>>>>>  button to indicate that they are ready to speak, although other
>>>>>  mechanism (e.g.  voice activation) are occasionally used.
>>>>>
>>>>>  Support for the Walkie-Talkie service can be deduced by 
>>>>>observing the
>>>>>  presence of a contact URI with a scheme of "sip:", 
>>>>
>>>>associated with
>>>>
>>>>
>>>>>  the following minimal set of capabilities: <audio>true</audio>
>>>>>  <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>>>>
>>>>>Presumably this is the same as the service sometimes called 
>>>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
>>>>>are several _non-interworking_ variants of this service, that 
>>>>>still all would match the characteristics listed above, as 
>>>>>the _SIP_ part might still be standard, at least for some 
>>>>>methods etc. But still the communication would fail even if 
>>>>>the tuple looks OK.
>>>>
>>>>I don't think I understand you here. I would expect that if 
>>>>both end-point
>>>>SIP US support exactly the same methods, the SIP UA should 
>>>>also be able to
>>>>handle those SIP method. Or, to phase it in another way, both 
>>>>end-point
>>>>should be able to communicate with each other only based on 
>>>>using these
>>>>capabilities.
>>>>
>>>
>>>
>>>To some extent that might be possible, but there might be differences e.g. in floor control etc., which make the service still practically unusable, which means that the presence document shouldn't give the impression that the communication will work. Obviously it is possible to list all those additional characteristics as well (and this might be a good idea too), but for instance if presence of a network game is published, it is easier to _explicitely_ tell what the game is by providing some sort of unique id which would be only meaningful to those similar game clients.
>>
>> >
>>
>>>>>(Also if I saw the characteristics above, 
>>>>>I could as well determine that they describe normal VoIP 
>>>>>application running in (very) some old PC with half-duplex 
>>>>>audio drivers, so I claim the service description is hard to 
>>>>>make unambiguous in the first place.)  
>>>>
>>I think the goal should be that all of these are at least minimally 
>>interoperable. I ought to be able to call a PTT contact from a regular 
>>voice phone and have *something* reasonable happen. (Maybe it is just 
>>oneway voice, with the non-ptt caller never getting the floor, but that 
>>may be better than nothing.
>>
>>I think the real problem in this case is not with the capability 
>>approach. It is that the 'duplex' capability is underspecified. There is 
>>no explanation of how a pair of half duplex endpoints (or one 
>>half-duplex and one full-duplex endpoint) agree on who can talk.
>>
>>In the case of the 'voice' capability, it is understood that having both 
>>endpoints support voice isn't enough to support communication. There is 
>>an SDP negotiation of codecs, etc. that either succeeds or fails. There 
>>should be something similar for half-duplex.
>>
>>
>>>>I also have the impression that it will be difficult to determine the
>>>>services merely based on the SIP methods (and ...) supported 
>>>>by the SIP UA.
>>>>It possibly might mean that the SIP-based ptt-service on my 
>>>>mobile phone
>>>>might be mapped onto some communication service on your PC. 
>>>>Then, I wonder
>>>>who will be responsible for such kind of mapping? Only the 
>>>>end terminals, or
>>>>is support from the network, presence server, necessary? And, 
>>>>how will one
>>>>service in one domain be mapped into a service in another 
>>>>domain? I believe
>>>>there are some open issues here. 
>>>
>>The intent is that capabilities have common meanings in all domains, so 
>>they don't need to be mapped. I don't see why there should be any 
>>binding between services and domains.
>>
>>In the general case one should assume that there are devices of widely 
>>varying capabilities in every domain, and that they know best what they 
>>are capable of. So the device should be the one deciding if a contact 
>>with and advertised set of capabilties is something it wants to attempt 
>>communication with.
>>
>>In some environment where there is an intermediary in front of a device 
>>that understands the device and acts on its behalf, then I suppose it 
>>could make these decisions. But I would consider that the odd case.
>>
>>
>>>I think that it is fully OK to describe some rather simple things using characteristics, but in many cases applications are more complex.
>>
>>Clearly we have chosen not to describe capabilities if full detail. 
>>(E.g. we don't publish codec support.) Because of this, there is the 
>>possibility that we make a wrong decision about ability to communicate 
>>and it doesn't get discovered until we try.
>>
>>This is ok when the expected probability of failure is low. (We 
>>generally expect that there will be some codec in common, or that one 
>>end or the other will be able to introduce a transcoder.)
>>
>>
>>>>>I think the main point of a service tuple is to give the 
>>>>>watcher enough information to know whether he can really 
>>>>>communicate & interoperate with the advertised service. Given 
>>>>>that many services taht are using SIP also have propritary or 
>>>>>non-SIP features, I don't think the current approach is enough. 
>>>>
>>In cases where the existing capabilities aren't sufficient to ensure a 
>>good probability of communication, then we should indeed consider adding 
>>additional capabilities.
>>
>>But I continue to object to the notion of "service" as the way to do 
>>that, because it conveys no semantics about what the capability is that 
>>it describes.
>>
>>
>>>>Do you have an example of a service that uses SIP and also has non
>>>>SIP-features? If your service also uses non-SIP features, you can not
>>>>determine the type of service by only looking at the SIP capabilities
>>>>supported by the SIP UA. Some indication of that non-SIP 
>>>>feature should also
>>>>be present in the presence document. 
>>>>
>>>
>>>
>>>True, and since some of those non-SIP features might not be based on (IETF) standards, it might be good to have the possibility for the developers of such applications to have a short cut of saying what the app really is, rather than having to list characteristics. In the end, some applications are really not even meant to work with any other than the exactly same application, for instance some networking games. 
>>
>>If the intent is to define a closed world, where interoperability is not 
>>even considered a desirable feature, then I agree. I think that may 
>>indeed be the case for proprietary games.
>>
>>But for something like PTT, while it might be in the best interest of 
>>the maker of a particular PTT solution to keep it closed, I don't think 
>>it is in the best interest of society or the IETF that this happen. This 
>>is precisely the forum in which we try to prevent that from happening.
>>
>>
>>>>>Of course each of these proprietary services are allowed to 
>>>>>define additional status or tuple-level extensions to PIDF.
>>>>
>>>>Yes, you can always extend the PIDF. My comment here is that 
>>>>this might lead
>>>>to an "explosion" of applications or services. It will 
>>>>certainly not help
>>>>interoperability.
>>>
>>My point precisely.
>>
>>
>>>And who would standardize such characteristic definitions. It has already taken a couple of years to standardize even "prescaps", which is just a starting point. 
>>>
>>>
>>>
>>>>>However, I would like to have some kind of "service 
>>>>>identifier" as part of the basic framework so that this could 
>>>>>be done in a consistent manner. I think it would help 
>>>>>especially in making of authorization and composition rules 
>>>>>more simple. The current "class" attribute is way too 
>>>>
>>>>loosely defined.
>>>>
>>>>
>>>>>So the concrete proposal is to include in RPID a "service id" 
>>>>>element that would have a vendor-specific namespace, similar 
>>>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
>>>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
>>>>><service-id>com.xyz.ptt</service-id>
>>>>>while the PTT app compliant with OMA would have, e.g.
>>>>><service-id>com.openmobilealliance.ptt</service-id>
>>>>
>>I am not fond of this solution. Done this way, we have defined an 
>>element that has no semantics at all, only syntax.
>>
>>I would rather see vendor specific extensions to PIDF. At least then the 
>>people that support an extension will understand what it means.
>>
>>But I would rather see people come forward and do the hard work of 
>>publicly defining new capabilities so that independent implementations 
>>can use them in an interoperable way.
>>
>>
>>>>In general, I could agree with that approach. However, I 
>>>>don't see a reason
>>>>why the vendor specific SIP-based PTT should be different 
>>>
>>>>from the OMA based
>>>
>>>>PTT. Simply define one PTT, based on the SIP UA capabilities.
>>>>
>>>
>>>
>>>Well, for instance Nokia has a prorietary PTT application, which uses SIP INVITE etc., so users are essentially addressed with sip: URI. However, it has some floor control, media etc. features which make it basically non-interoperable with clients that are not specifically following that proprietary spec. 
>>
>>This is a problem, not a good thing. I realize it has to happen as part 
>>of the development process, but the goal should be to get out of it as 
>>fast as possible.
>>
>>I would hope that endpoints supporting this would be described as well 
>>as possible using existing capabilities, and that a reasonable effort be 
>>made to do something reasonable when some other sip device connects.
>>
>>If you really can't deal with a regular voice caller at all, then I 
>>would suggest, as a transitional step, saying that you don't support 
>>voice, and then advertising some proprietary capability as an 
>>alternative to voice.
>>
>>
>>>OMA PTT is closer to standard IETF SIP. Unless there is some conversion, it does not work with Nokia PTT directly. There should be some way for distinguishing these two in a presence document, so that there won't be confusion. I think the current idea is to do some kind of proprietary PIDF extensions, but in my opinion a standard XML element in RPID to tell this kind of information would be better, also for the purposes of authorization rule making.
>>
>>See above for what I think you could do for now. The same issues may 
>>apply to OMA PTT, and a similar solution could be applied to it.
>>
>>Hopefully somebody will come up with a PTT that can be dealt with more 
>>gracefully. I think that work should be done on better defining 
>>half-duplex, so that it can be used in conjunction with voice to 
>>describe PTT. But it may just turn out to be the wrong thing. Maybe we 
>>need a "floor-control" capability.
>>
>>
>>>>>in _addition_ to describing sip:, half-duplex audio, and the 
>>>>>supported methods. Now:
>>>>>1.) Watcher having one of those apps could see right away 
>>>>>whether it is possible to communicate
>>>>>2.) Composer getting PUBLISHes from 2 sources could 
>>>>>immediately know that they are from a similar app
>>>>>3.) The app could set its authorization rules using the unique id.
>>>>>
>>>>>The main downside I can see in this approach that if there 
>>>>>are two different proprietary apps that indeed could 
>>>>>communicate at least partially, the watcher might make a 
>>>>>conclusion that communication is not possible baed on 
>>>>>different service-id.
>>>>
>>This is my concern as well.
>>
>> >>> However, in this case the watcher still
>>
>>>>>would have the characteristics visible and could determine 
>>>>>that some interworking could be achieved. 
>>>>
>>>>Personally, I don't think it is a good idea that an end-user 
>>>>should make
>>>>such decission. If I get a presence document that indicates 
>>>>that another
>>>>person is able to receive some type of communication, I 
>>>>expect that that
>>>>type of communication would actually work when I use it.
>>>
>>With high probability. There are no guarantees with presence. For 
>>instance see my mention above of codecs.
>>
>>
>>>Exactly right. If you are an application who does not necessarily aim for interoperability, but still uses sip:, INVITE etc., you should be able to explicitely say so.
>>
>>And you can, without a special notion of service.
>>
>>
>>>>>For this kind of 
>>>>>reasons there should be careful text about when to use 
>>>>>service-id in the first place, and what can be determined from it.
>>>>>
>>>>>Does someone think that this is an absolutely bad idea? If 
>>>>>so, how would you envision solving the issues discussed above?
>>>>
>>I still don't buy this.
>>
>>
>>>>I think more care should be taken in mapping SIP UA 
>>>>capabilities into actual
>>>>services. That is not so clear to me!
>>>
>>>The idea is nice, but I don't see a problem in _addition_ defining the application with something like "com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind of profile would have more meaning than the list of characteristics.
>>
>>I am still opposed to defining an attribute with no semantics.
>>We have been around this so many times that I am quite certain that it 
>>will never be possible to get a useful agreement on what "service" means.
>>
>>If it is introduced, I expect it to have problems like those of the INFO 
>>method. (The arguments for introducing it seem similar.)
>>
>>	Paul
>>
>>
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From simple-bounces@ietf.org  Fri Sep 24 10:27:26 2004
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Date: Fri, 24 Sep 2004 10:17:59 -0400
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Subject: Re: [Simple] Presence Data Model: Identifying services
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Goeman Stefan wrote:
> Hello All,
> 
> I would like to add the following questions/problems to the discussion.
> 
> How will a presence document be presented to an end-user. Personally, I
> don't think it makes much sense to present a "raw" presence document, like
> the one on page 26 of the Presence Data Model draft, to the end-user.

Agreed. I don't know of anybody that thinks it should normally be 
displayed in raw form. (For advanced users, a device with a sufficiently 
powerful UI could offer something on the order of the "view source" that 
mail guis often provide.)

 > In the
> end, people would like to see on their device: "I can contact person X via
> email, voice-call, IM, ...". I don't think they would like to see things
> like: "I can contact person X via SIP". I believe that 99.99% of the world
> does not know what SIP is, yet everybody (well ...) knows what email is. 

Everybody I have spoken to about this seems to imagine some form of 
iconic interface - similar to but richer than what is provided by common 
IM clients today for buddy lists.

This will of course vary somewhat depending on the capabilities of the 
client device. For a device with the ui capabilities of a PC, running a 
client application with multimedia support (voice, video, IM, ...), what 
I have in mind is something like:

- a "list" view with one line per buddy
- the list has an icons indicating general availability, idleness,
   supported media. (This represents an OR of all the tuples.)
- this list should have been pruned by the client application, so
   that it only displays capabilities the client also supports,
   and only indicated general availability if some supported mode
   of communication is available.
- it would be possible to initiate a call of any supported type(s)
   by interacting with this single line on the display. The client
   would then presumably "fork" the call to all compatible tuples.
- it should be possible to "expand" the display of a buddy.
   In the expanded form, there would be one line per tuple.
   Again these would be iconic. A call to a particular device could
   be initiated by manipulating one of these.

> The point I want to make is that some entity has to process a presence
> document in a way that it can be presented to an end-user in an meaningful
> manner. 

Yes, filter and format.

> My first guess was that a device could the transformation of the "raw"
> presence document. Device know which applications they are currently
> running. So, there should be no problem here.
> 
> On second thought, I realized that there is one problem:
> How will this work together with the Authorization rules.

I think you are imagining a problem here. I don't think it makes much 
sense for the client to determine authorization, beyond simply being 
authorized to subscribe to presence. If the PS gives you the info, just 
display it. The PS can filter out what it thinks you should not see or do.

In any case there is no good way for a client to determine if it is 
permitted to do something, short of trying to do it.

	Paul

 > Here, again, you
> have to present the authorization rules in a meaningful manner (via some
> front end user interface). If I am correct, the idea is to retrieve the
> authorization rules via XCAP from the policy server, make some changes and
> provide the changes back to the policy server via XCAP.  Yet, the policy
> server is not aware of any services like IM, voice call, ... . So, again,
> the device should make this mapping. This works well if you only use one
> device. But, if you use more than one device, with different capabilities
> and services, it does not work anymore. A device can't say something
> meaningful over a service it does not support. Consequently, you can not
> edit every aspect of the authorization rules on your device. I think this is
> problematic, and I don't see a good solution right know.
> 
> Greetings,
> Stefan. 
> 
> 
>>-----Original Message-----
>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>>Sent: donderdag 23 september 2004 17:47
>>To: Markus.Isomaki@nokia.com
>>Cc: Goeman Stefan; simple@ietf.org
>>Subject: Re: [Simple] Presence Data Model: Identifying services
>>
>>
>>I have comments both to Markus' original posting and to the ensuing 
>>discussion. So I will just insert inline in the latest 
>>response I have.
>>
>>	Paul
>>
>>Markus.Isomaki@nokia.com wrote:
>>
>>>Hi Stefan,
>>>
>>>Comments back to you inline. 
>>>
>>>(Yes, I absolutely agree with you on the usefulness of the 
>>
>>Presence Data Model draft and that it should have been the 
>>starting point, at least right after PIDF.)
>>
>>>
>>>>-----Original Message-----
>>>>From: simple-bounces@ietf.org 
>>>>[mailto:simple-bounces@ietf.org]On Behalf
>>>>Of ext Goeman Stefan
>>>>Sent: 23 September, 2004 14:42
>>>>To: 'simple@ietf.org'
>>>>Subject: RE: [Simple] Presence Data Model: Identifying services
>>>>
>>>>
>>>>Hello,
>>>>
>>>>Comments inline
>>>>
>>>>P.S.: In general, I find the Presence Data model draft a good 
>>>>document. This
>>>>is how things should have started in the first place.
>>>>
>>>>Greetings,
>>>>Stefan.
>>>>
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: simple-bounces@ietf.org 
>>>>>[mailto:simple-bounces@ietf.org] On Behalf Of 
>>>>
>>>>Markus.Isomaki@nokia.com
>>>>
>>>>
>>>>>Sent: donderdag 23 september 2004 10:10
>>>>>To: simple@ietf.org
>>>>>Subject: [Simple] Presence Data Model: Identifying services
>>>>>
>>>>>
>>>>>Hi all,
>>>>>
>>>>>I have a comment on Presence Data Model draft about the 
>>>>>identification/characterization of services. (I know this has 
>>>>>been discussed for a while already, but I still think the 
>>>>>current way is somewhat unadequate.)
>>>>>
>>>>>The current Data Model draft says:
>>>>>
>>>>>  In this model, services are not explicitly enumerated.  That is,
>>>>>  there is no "service" attribute with values of "ptt" or 
>>>>>"telephony".
>>>>>  Rather, the service is identified in one of two ways.  In 
>>>>>many cases,
>>>>>  the URI scheme is a clear indicator of service.  An "sms" 
>>>>>URI clearly
>>>>>  indicates SMS, and a "tel" URI clearly indicates 
>>>>>telephony. 
>>>>
>>I don't find the scheme as clearcut as that. I don't think one should 
>>assume "tel" indicates telephony. It perhaps does imply 
>>reachability via 
>>various telephony protocols for purpose of voice 
>>communications, but is 
>>not exclusive to that. If the tel: URI can be mapped to a sip 
>>endpoint 
>>and protocol, then non-voice, non-telephony capabilities are 
>>not excluded.
>>
>>For instance, it may prove convenient to publish a tel URI for a sip 
>>endpoint that is capable of both voice and IM. Then calls via 
>>the PSTN 
>>will work, calls from a sip audio phone will work, calls from another 
>>voice+im client will have access to voice and IM.
>>
>> >>> For some
>>
>>>>>  URIs, there may be many services available, for example, 
>>>>
>>>>SIP.  For
>>>>
>>>>
>>>>>  those services, each service has a set of 
>>>>
>>>>characteristics, each of
>>>>
>>>>
>>>>>  which has a well-defined meaning, such that a system can
>>>>>  unequivocally determine whether or not the service has that
>>>>>  characteristic.  This is discussed in more detail in [5]. 
>>>>>
>>>>>I agree that the contact URI scheme is the most important 
>>>>>piece of information to distinguish what is the "service". 
>>>>>(There are some gaps here too, since e.g. the URI schemes for 
>>>>>SMS or MMS do not even exist, and there may be other non-IETF 
>>>>>protocols that face the same problem.) However, I'm not 
>>>>>convinced that listing the signaling/media characteristics of 
>>>>>the end-point or service really gives enough information to 
>>>>>the watcher to really determine if it can communicate with 
>>>>>the advertised service/application. 
>>>>>
>>>>>The refenced draft [5] has a following example:
>>>>>
>>>>>5.6 Walkie-talkie
>>>>>
>>>>>  The walkie-talkie service allows real-time voice communication
>>>>>  between participants.  Only one participant can speak at a 
>>>>>time; that
>>>>>  is, communication is half-duplex.  Typically, 
>>>>
>>>>participants press a
>>>>
>>>>
>>>>>  button to indicate that they are ready to speak, although other
>>>>>  mechanism (e.g.  voice activation) are occasionally used.
>>>>>
>>>>>  Support for the Walkie-Talkie service can be deduced by 
>>>>>observing the
>>>>>  presence of a contact URI with a scheme of "sip:", 
>>>>
>>>>associated with
>>>>
>>>>
>>>>>  the following minimal set of capabilities: <audio>true</audio>
>>>>>  <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>>>>
>>>>>Presumably this is the same as the service sometimes called 
>>>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
>>>>>are several _non-interworking_ variants of this service, that 
>>>>>still all would match the characteristics listed above, as 
>>>>>the _SIP_ part might still be standard, at least for some 
>>>>>methods etc. But still the communication would fail even if 
>>>>>the tuple looks OK.
>>>>
>>>>I don't think I understand you here. I would expect that if 
>>>>both end-point
>>>>SIP US support exactly the same methods, the SIP UA should 
>>>>also be able to
>>>>handle those SIP method. Or, to phase it in another way, both 
>>>>end-point
>>>>should be able to communicate with each other only based on 
>>>>using these
>>>>capabilities.
>>>>
>>>
>>>
>>>To some extent that might be possible, but there might be 
>>
>>differences e.g. in floor control etc., which make the 
>>service still practically unusable, which means that the 
>>presence document shouldn't give the impression that the 
>>communication will work. Obviously it is possible to list all 
>>those additional characteristics as well (and this might be a 
>>good idea too), but for instance if presence of a network 
>>game is published, it is easier to _explicitely_ tell what 
>>the game is by providing some sort of unique id which would 
>>be only meaningful to those similar game clients.
>> >
>>
>>>>>(Also if I saw the characteristics above, 
>>>>>I could as well determine that they describe normal VoIP 
>>>>>application running in (very) some old PC with half-duplex 
>>>>>audio drivers, so I claim the service description is hard to 
>>>>>make unambiguous in the first place.)  
>>>>
>>I think the goal should be that all of these are at least minimally 
>>interoperable. I ought to be able to call a PTT contact from 
>>a regular 
>>voice phone and have *something* reasonable happen. (Maybe it is just 
>>oneway voice, with the non-ptt caller never getting the 
>>floor, but that 
>>may be better than nothing.
>>
>>I think the real problem in this case is not with the capability 
>>approach. It is that the 'duplex' capability is 
>>underspecified. There is 
>>no explanation of how a pair of half duplex endpoints (or one 
>>half-duplex and one full-duplex endpoint) agree on who can talk.
>>
>>In the case of the 'voice' capability, it is understood that 
>>having both 
>>endpoints support voice isn't enough to support 
>>communication. There is 
>>an SDP negotiation of codecs, etc. that either succeeds or 
>>fails. There 
>>should be something similar for half-duplex.
>>
>>
>>>>I also have the impression that it will be difficult to 
>>>
>>determine the
>>
>>>>services merely based on the SIP methods (and ...) supported 
>>>>by the SIP UA.
>>>>It possibly might mean that the SIP-based ptt-service on my 
>>>>mobile phone
>>>>might be mapped onto some communication service on your PC. 
>>>>Then, I wonder
>>>>who will be responsible for such kind of mapping? Only the 
>>>>end terminals, or
>>>>is support from the network, presence server, necessary? And, 
>>>>how will one
>>>>service in one domain be mapped into a service in another 
>>>>domain? I believe
>>>>there are some open issues here. 
>>>
>>The intent is that capabilities have common meanings in all 
>>domains, so 
>>they don't need to be mapped. I don't see why there should be any 
>>binding between services and domains.
>>
>>In the general case one should assume that there are devices 
>>of widely 
>>varying capabilities in every domain, and that they know best 
>>what they 
>>are capable of. So the device should be the one deciding if a contact 
>>with and advertised set of capabilties is something it wants 
>>to attempt 
>>communication with.
>>
>>In some environment where there is an intermediary in front 
>>of a device 
>>that understands the device and acts on its behalf, then I suppose it 
>>could make these decisions. But I would consider that the odd case.
>>
>>
>>>I think that it is fully OK to describe some rather simple 
>>
>>things using characteristics, but in many cases applications 
>>are more complex.
>>
>>Clearly we have chosen not to describe capabilities if full detail. 
>>(E.g. we don't publish codec support.) Because of this, there is the 
>>possibility that we make a wrong decision about ability to 
>>communicate 
>>and it doesn't get discovered until we try.
>>
>>This is ok when the expected probability of failure is low. (We 
>>generally expect that there will be some codec in common, or that one 
>>end or the other will be able to introduce a transcoder.)
>>
>>
>>>>>I think the main point of a service tuple is to give the 
>>>>>watcher enough information to know whether he can really 
>>>>>communicate & interoperate with the advertised service. Given 
>>>>>that many services taht are using SIP also have propritary or 
>>>>>non-SIP features, I don't think the current approach is enough. 
>>>>
>>In cases where the existing capabilities aren't sufficient to 
>>ensure a 
>>good probability of communication, then we should indeed 
>>consider adding 
>>additional capabilities.
>>
>>But I continue to object to the notion of "service" as the way to do 
>>that, because it conveys no semantics about what the 
>>capability is that 
>>it describes.
>>
>>
>>>>Do you have an example of a service that uses SIP and also has non
>>>>SIP-features? If your service also uses non-SIP features, 
>>>
>>you can not
>>
>>>>determine the type of service by only looking at the SIP 
>>>
>>capabilities
>>
>>>>supported by the SIP UA. Some indication of that non-SIP 
>>>>feature should also
>>>>be present in the presence document. 
>>>>
>>>
>>>
>>>True, and since some of those non-SIP features might not be 
>>
>>based on (IETF) standards, it might be good to have the 
>>possibility for the developers of such applications to have a 
>>short cut of saying what the app really is, rather than 
>>having to list characteristics. In the end, some applications 
>>are really not even meant to work with any other than the 
>>exactly same application, for instance some networking games. 
>>
>>If the intent is to define a closed world, where 
>>interoperability is not 
>>even considered a desirable feature, then I agree. I think that may 
>>indeed be the case for proprietary games.
>>
>>But for something like PTT, while it might be in the best interest of 
>>the maker of a particular PTT solution to keep it closed, I 
>>don't think 
>>it is in the best interest of society or the IETF that this 
>>happen. This 
>>is precisely the forum in which we try to prevent that from happening.
>>
>>
>>>>>Of course each of these proprietary services are allowed to 
>>>>>define additional status or tuple-level extensions to PIDF.
>>>>
>>>>Yes, you can always extend the PIDF. My comment here is that 
>>>>this might lead
>>>>to an "explosion" of applications or services. It will 
>>>>certainly not help
>>>>interoperability.
>>>
>>My point precisely.
>>
>>
>>>And who would standardize such characteristic definitions. 
>>
>>It has already taken a couple of years to standardize even 
>>"prescaps", which is just a starting point. 
>>
>>>
>>>>>However, I would like to have some kind of "service 
>>>>>identifier" as part of the basic framework so that this could 
>>>>>be done in a consistent manner. I think it would help 
>>>>>especially in making of authorization and composition rules 
>>>>>more simple. The current "class" attribute is way too 
>>>>
>>>>loosely defined.
>>>>
>>>>
>>>>>So the concrete proposal is to include in RPID a "service id" 
>>>>>element that would have a vendor-specific namespace, similar 
>>>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
>>>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
>>>>><service-id>com.xyz.ptt</service-id>
>>>>>while the PTT app compliant with OMA would have, e.g.
>>>>><service-id>com.openmobilealliance.ptt</service-id>
>>>>
>>I am not fond of this solution. Done this way, we have defined an 
>>element that has no semantics at all, only syntax.
>>
>>I would rather see vendor specific extensions to PIDF. At 
>>least then the 
>>people that support an extension will understand what it means.
>>
>>But I would rather see people come forward and do the hard work of 
>>publicly defining new capabilities so that independent 
>>implementations 
>>can use them in an interoperable way.
>>
>>
>>>>In general, I could agree with that approach. However, I 
>>>>don't see a reason
>>>>why the vendor specific SIP-based PTT should be different 
>>>
>>>>from the OMA based
>>>
>>>>PTT. Simply define one PTT, based on the SIP UA capabilities.
>>>>
>>>
>>>
>>>Well, for instance Nokia has a prorietary PTT application, 
>>
>>which uses SIP INVITE etc., so users are essentially 
>>addressed with sip: URI. However, it has some floor control, 
>>media etc. features which make it basically non-interoperable 
>>with clients that are not specifically following that 
>>proprietary spec. 
>>
>>This is a problem, not a good thing. I realize it has to 
>>happen as part 
>>of the development process, but the goal should be to get out 
>>of it as 
>>fast as possible.
>>
>>I would hope that endpoints supporting this would be 
>>described as well 
>>as possible using existing capabilities, and that a 
>>reasonable effort be 
>>made to do something reasonable when some other sip device connects.
>>
>>If you really can't deal with a regular voice caller at all, then I 
>>would suggest, as a transitional step, saying that you don't support 
>>voice, and then advertising some proprietary capability as an 
>>alternative to voice.
>>
>>
>>>OMA PTT is closer to standard IETF SIP. Unless there is 
>>
>>some conversion, it does not work with Nokia PTT directly. 
>>There should be some way for distinguishing these two in a 
>>presence document, so that there won't be confusion. I think 
>>the current idea is to do some kind of proprietary PIDF 
>>extensions, but in my opinion a standard XML element in RPID 
>>to tell this kind of information would be better, also for 
>>the purposes of authorization rule making.
>>
>>See above for what I think you could do for now. The same issues may 
>>apply to OMA PTT, and a similar solution could be applied to it.
>>
>>Hopefully somebody will come up with a PTT that can be dealt 
>>with more 
>>gracefully. I think that work should be done on better defining 
>>half-duplex, so that it can be used in conjunction with voice to 
>>describe PTT. But it may just turn out to be the wrong thing. 
>>Maybe we 
>>need a "floor-control" capability.
>>
>>
>>>>>in _addition_ to describing sip:, half-duplex audio, and the 
>>>>>supported methods. Now:
>>>>>1.) Watcher having one of those apps could see right away 
>>>>>whether it is possible to communicate
>>>>>2.) Composer getting PUBLISHes from 2 sources could 
>>>>>immediately know that they are from a similar app
>>>>>3.) The app could set its authorization rules using the unique id.
>>>>>
>>>>>The main downside I can see in this approach that if there 
>>>>>are two different proprietary apps that indeed could 
>>>>>communicate at least partially, the watcher might make a 
>>>>>conclusion that communication is not possible baed on 
>>>>>different service-id.
>>>>
>>This is my concern as well.
>>
>> >>> However, in this case the watcher still
>>
>>>>>would have the characteristics visible and could determine 
>>>>>that some interworking could be achieved. 
>>>>
>>>>Personally, I don't think it is a good idea that an end-user 
>>>>should make
>>>>such decission. If I get a presence document that indicates 
>>>>that another
>>>>person is able to receive some type of communication, I 
>>>>expect that that
>>>>type of communication would actually work when I use it.
>>>
>>With high probability. There are no guarantees with presence. For 
>>instance see my mention above of codecs.
>>
>>
>>>Exactly right. If you are an application who does not 
>>
>>necessarily aim for interoperability, but still uses sip:, 
>>INVITE etc., you should be able to explicitely say so.
>>
>>And you can, without a special notion of service.
>>
>>
>>>>>For this kind of 
>>>>>reasons there should be careful text about when to use 
>>>>>service-id in the first place, and what can be determined from it.
>>>>>
>>>>>Does someone think that this is an absolutely bad idea? If 
>>>>>so, how would you envision solving the issues discussed above?
>>>>
>>I still don't buy this.
>>
>>
>>>>I think more care should be taken in mapping SIP UA 
>>>>capabilities into actual
>>>>services. That is not so clear to me!
>>>
>>>The idea is nice, but I don't see a problem in _addition_ 
>>
>>defining the application with something like 
>>"com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind 
>>of profile would have more meaning than the list of characteristics.
>>
>>I am still opposed to defining an attribute with no semantics.
>>We have been around this so many times that I am quite 
>>certain that it 
>>will never be possible to get a useful agreement on what 
>>"service" means.
>>
>>If it is introduced, I expect it to have problems like those 
>>of the INFO 
>>method. (The arguments for introducing it seem similar.)
>>
>>	Paul
>>
> 
> 
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From simple-bounces@ietf.org  Fri Sep 24 10:40:07 2004
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Helsen,

I don't understand your terminology very well, or what you are trying to 
achieve.

By 'client' I guess you mean a device addressed by a contact in a tuple 
of the presence document, rather than a subscriber to presence. Is that 
right?

I don't know precisely what role you have in mind for 'operator'. Is it 
the operator of the presence server itself? Or is it something that is a 
client of the PS. What is it doing that requires it to know this 
information?

In general the device will probably be publishing its own presence 
status to the presence server. If so the extension you suggest would 
simply cause it to push this invariant information over and over.

This kind of info can already be polled using OPTIONS. Why isn't that 
enough?

	Paul

Helsen Frank wrote:
> Hello,
> 
> 
> At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a way 
> to indicate to the server which client the user uses.  I can imagine 
> that an operator has specific client capabilities he would like to know 
> (e.g. java virtual machine version, screen size).  When including this 
> client information, the server could retrieve the client capabilities 
> from his e.g. his DB.  With this extension, the operator has control 
> over the client information and is not dependend on the information the 
> client provides to him. 
> 
> Next example shows a possible extension
> 
> <ClientInfo>
>         <Manufacturor> Siemens </Manufacturor>
>         <Model> CX65 </Model>
>         <Version>2.103564</Version>
> </ClientInfo>
> 
> Regards,
> 
> Frank
> 
> 
> 
> ------------------------------------------------------------------------
> 
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From simple-bounces@ietf.org  Fri Sep 24 10:42:02 2004
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From: Goeman Stefan <Stefan.Goeman@siemens.com>
To: "'simple@ietf.org'" <simple@ietf.org>
Subject: RE: [Simple] Presence Data Model: Identifying services
Date: Fri, 24 Sep 2004 16:31:07 +0200
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Hello,

Comments inline.

Greetings,
Stefan.

> > On second thought, I realized that there is one problem:
> > How will this work together with the Authorization rules.
> 
> I think you are imagining a problem here. I don't think it makes much 
> sense for the client to determine authorization, beyond simply being 
> authorized to subscribe to presence. If the PS gives you the 
> info, just 
> display it. The PS can filter out what it thinks you should 
> not see or do.
>
I think we are talking about different things here. I was taking about me as
presentity managing and editing my own authorization rules. Since my
presence information is also "my property" I would expect that I would be
able to play the role of a rule maker. Clearly, you have to edit your rules
somehow, via some device.
 
> In any case there is no good way for a client to determine if it is 
> permitted to do something, short of trying to do it.
> 
> 	Paul
> 
>  > Here, again, you
> > have to present the authorization rules in a meaningful 
> manner (via some
> > front end user interface). If I am correct, the idea is to 
> retrieve the
> > authorization rules via XCAP from the policy server, make 
> some changes and
> > provide the changes back to the policy server via XCAP.  
> Yet, the policy
> > server is not aware of any services like IM, voice call, 
> ... . So, again,
> > the device should make this mapping. This works well if you 
> only use one
> > device. But, if you use more than one device, with 
> different capabilities
> > and services, it does not work anymore. A device can't say something
> > meaningful over a service it does not support. 
> Consequently, you can not
> > edit every aspect of the authorization rules on your 
> device. I think this is
> > problematic, and I don't see a good solution right know.
> > 
> > Greetings,
> > Stefan. 
> > 
> > 
> >>-----Original Message-----
> >>From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> >>Sent: donderdag 23 september 2004 17:47
> >>To: Markus.Isomaki@nokia.com
> >>Cc: Goeman Stefan; simple@ietf.org
> >>Subject: Re: [Simple] Presence Data Model: Identifying services
> >>
> >>
> >>I have comments both to Markus' original posting and to the ensuing 
> >>discussion. So I will just insert inline in the latest 
> >>response I have.
> >>
> >>	Paul
> >>
> >>Markus.Isomaki@nokia.com wrote:
> >>
> >>>Hi Stefan,
> >>>
> >>>Comments back to you inline. 
> >>>
> >>>(Yes, I absolutely agree with you on the usefulness of the 
> >>
> >>Presence Data Model draft and that it should have been the 
> >>starting point, at least right after PIDF.)
> >>
> >>>
> >>>>-----Original Message-----
> >>>>From: simple-bounces@ietf.org 
> >>>>[mailto:simple-bounces@ietf.org]On Behalf
> >>>>Of ext Goeman Stefan
> >>>>Sent: 23 September, 2004 14:42
> >>>>To: 'simple@ietf.org'
> >>>>Subject: RE: [Simple] Presence Data Model: Identifying services
> >>>>
> >>>>
> >>>>Hello,
> >>>>
> >>>>Comments inline
> >>>>
> >>>>P.S.: In general, I find the Presence Data model draft a good 
> >>>>document. This
> >>>>is how things should have started in the first place.
> >>>>
> >>>>Greetings,
> >>>>Stefan.
> >>>>
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: simple-bounces@ietf.org 
> >>>>>[mailto:simple-bounces@ietf.org] On Behalf Of 
> >>>>
> >>>>Markus.Isomaki@nokia.com
> >>>>
> >>>>
> >>>>>Sent: donderdag 23 september 2004 10:10
> >>>>>To: simple@ietf.org
> >>>>>Subject: [Simple] Presence Data Model: Identifying services
> >>>>>
> >>>>>
> >>>>>Hi all,
> >>>>>
> >>>>>I have a comment on Presence Data Model draft about the 
> >>>>>identification/characterization of services. (I know this has 
> >>>>>been discussed for a while already, but I still think the 
> >>>>>current way is somewhat unadequate.)
> >>>>>
> >>>>>The current Data Model draft says:
> >>>>>
> >>>>>  In this model, services are not explicitly enumerated. 
>  That is,
> >>>>>  there is no "service" attribute with values of "ptt" or 
> >>>>>"telephony".
> >>>>>  Rather, the service is identified in one of two ways.  In 
> >>>>>many cases,
> >>>>>  the URI scheme is a clear indicator of service.  An "sms" 
> >>>>>URI clearly
> >>>>>  indicates SMS, and a "tel" URI clearly indicates 
> >>>>>telephony. 
> >>>>
> >>I don't find the scheme as clearcut as that. I don't think 
> one should 
> >>assume "tel" indicates telephony. It perhaps does imply 
> >>reachability via 
> >>various telephony protocols for purpose of voice 
> >>communications, but is 
> >>not exclusive to that. If the tel: URI can be mapped to a sip 
> >>endpoint 
> >>and protocol, then non-voice, non-telephony capabilities are 
> >>not excluded.
> >>
> >>For instance, it may prove convenient to publish a tel URI 
> for a sip 
> >>endpoint that is capable of both voice and IM. Then calls via 
> >>the PSTN 
> >>will work, calls from a sip audio phone will work, calls 
> from another 
> >>voice+im client will have access to voice and IM.
> >>
> >> >>> For some
> >>
> >>>>>  URIs, there may be many services available, for example, 
> >>>>
> >>>>SIP.  For
> >>>>
> >>>>
> >>>>>  those services, each service has a set of 
> >>>>
> >>>>characteristics, each of
> >>>>
> >>>>
> >>>>>  which has a well-defined meaning, such that a system can
> >>>>>  unequivocally determine whether or not the service has that
> >>>>>  characteristic.  This is discussed in more detail in [5]. 
> >>>>>
> >>>>>I agree that the contact URI scheme is the most important 
> >>>>>piece of information to distinguish what is the "service". 
> >>>>>(There are some gaps here too, since e.g. the URI schemes for 
> >>>>>SMS or MMS do not even exist, and there may be other non-IETF 
> >>>>>protocols that face the same problem.) However, I'm not 
> >>>>>convinced that listing the signaling/media characteristics of 
> >>>>>the end-point or service really gives enough information to 
> >>>>>the watcher to really determine if it can communicate with 
> >>>>>the advertised service/application. 
> >>>>>
> >>>>>The refenced draft [5] has a following example:
> >>>>>
> >>>>>5.6 Walkie-talkie
> >>>>>
> >>>>>  The walkie-talkie service allows real-time voice communication
> >>>>>  between participants.  Only one participant can speak at a 
> >>>>>time; that
> >>>>>  is, communication is half-duplex.  Typically, 
> >>>>
> >>>>participants press a
> >>>>
> >>>>
> >>>>>  button to indicate that they are ready to speak, although other
> >>>>>  mechanism (e.g.  voice activation) are occasionally used.
> >>>>>
> >>>>>  Support for the Walkie-Talkie service can be deduced by 
> >>>>>observing the
> >>>>>  presence of a contact URI with a scheme of "sip:", 
> >>>>
> >>>>associated with
> >>>>
> >>>>
> >>>>>  the following minimal set of capabilities: <audio>true</audio>
> >>>>>  <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> >>>>>
> >>>>>Presumably this is the same as the service sometimes called 
> >>>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
> >>>>>are several _non-interworking_ variants of this service, that 
> >>>>>still all would match the characteristics listed above, as 
> >>>>>the _SIP_ part might still be standard, at least for some 
> >>>>>methods etc. But still the communication would fail even if 
> >>>>>the tuple looks OK.
> >>>>
> >>>>I don't think I understand you here. I would expect that if 
> >>>>both end-point
> >>>>SIP US support exactly the same methods, the SIP UA should 
> >>>>also be able to
> >>>>handle those SIP method. Or, to phase it in another way, both 
> >>>>end-point
> >>>>should be able to communicate with each other only based on 
> >>>>using these
> >>>>capabilities.
> >>>>
> >>>
> >>>
> >>>To some extent that might be possible, but there might be 
> >>
> >>differences e.g. in floor control etc., which make the 
> >>service still practically unusable, which means that the 
> >>presence document shouldn't give the impression that the 
> >>communication will work. Obviously it is possible to list all 
> >>those additional characteristics as well (and this might be a 
> >>good idea too), but for instance if presence of a network 
> >>game is published, it is easier to _explicitely_ tell what 
> >>the game is by providing some sort of unique id which would 
> >>be only meaningful to those similar game clients.
> >> >
> >>
> >>>>>(Also if I saw the characteristics above, 
> >>>>>I could as well determine that they describe normal VoIP 
> >>>>>application running in (very) some old PC with half-duplex 
> >>>>>audio drivers, so I claim the service description is hard to 
> >>>>>make unambiguous in the first place.)  
> >>>>
> >>I think the goal should be that all of these are at least minimally 
> >>interoperable. I ought to be able to call a PTT contact from 
> >>a regular 
> >>voice phone and have *something* reasonable happen. (Maybe 
> it is just 
> >>oneway voice, with the non-ptt caller never getting the 
> >>floor, but that 
> >>may be better than nothing.
> >>
> >>I think the real problem in this case is not with the capability 
> >>approach. It is that the 'duplex' capability is 
> >>underspecified. There is 
> >>no explanation of how a pair of half duplex endpoints (or one 
> >>half-duplex and one full-duplex endpoint) agree on who can talk.
> >>
> >>In the case of the 'voice' capability, it is understood that 
> >>having both 
> >>endpoints support voice isn't enough to support 
> >>communication. There is 
> >>an SDP negotiation of codecs, etc. that either succeeds or 
> >>fails. There 
> >>should be something similar for half-duplex.
> >>
> >>
> >>>>I also have the impression that it will be difficult to 
> >>>
> >>determine the
> >>
> >>>>services merely based on the SIP methods (and ...) supported 
> >>>>by the SIP UA.
> >>>>It possibly might mean that the SIP-based ptt-service on my 
> >>>>mobile phone
> >>>>might be mapped onto some communication service on your PC. 
> >>>>Then, I wonder
> >>>>who will be responsible for such kind of mapping? Only the 
> >>>>end terminals, or
> >>>>is support from the network, presence server, necessary? And, 
> >>>>how will one
> >>>>service in one domain be mapped into a service in another 
> >>>>domain? I believe
> >>>>there are some open issues here. 
> >>>
> >>The intent is that capabilities have common meanings in all 
> >>domains, so 
> >>they don't need to be mapped. I don't see why there should be any 
> >>binding between services and domains.
> >>
> >>In the general case one should assume that there are devices 
> >>of widely 
> >>varying capabilities in every domain, and that they know best 
> >>what they 
> >>are capable of. So the device should be the one deciding if 
> a contact 
> >>with and advertised set of capabilties is something it wants 
> >>to attempt 
> >>communication with.
> >>
> >>In some environment where there is an intermediary in front 
> >>of a device 
> >>that understands the device and acts on its behalf, then I 
> suppose it 
> >>could make these decisions. But I would consider that the odd case.
> >>
> >>
> >>>I think that it is fully OK to describe some rather simple 
> >>
> >>things using characteristics, but in many cases applications 
> >>are more complex.
> >>
> >>Clearly we have chosen not to describe capabilities if full detail. 
> >>(E.g. we don't publish codec support.) Because of this, 
> there is the 
> >>possibility that we make a wrong decision about ability to 
> >>communicate 
> >>and it doesn't get discovered until we try.
> >>
> >>This is ok when the expected probability of failure is low. (We 
> >>generally expect that there will be some codec in common, 
> or that one 
> >>end or the other will be able to introduce a transcoder.)
> >>
> >>
> >>>>>I think the main point of a service tuple is to give the 
> >>>>>watcher enough information to know whether he can really 
> >>>>>communicate & interoperate with the advertised service. Given 
> >>>>>that many services taht are using SIP also have propritary or 
> >>>>>non-SIP features, I don't think the current approach is enough. 
> >>>>
> >>In cases where the existing capabilities aren't sufficient to 
> >>ensure a 
> >>good probability of communication, then we should indeed 
> >>consider adding 
> >>additional capabilities.
> >>
> >>But I continue to object to the notion of "service" as the 
> way to do 
> >>that, because it conveys no semantics about what the 
> >>capability is that 
> >>it describes.
> >>
> >>
> >>>>Do you have an example of a service that uses SIP and also has non
> >>>>SIP-features? If your service also uses non-SIP features, 
> >>>
> >>you can not
> >>
> >>>>determine the type of service by only looking at the SIP 
> >>>
> >>capabilities
> >>
> >>>>supported by the SIP UA. Some indication of that non-SIP 
> >>>>feature should also
> >>>>be present in the presence document. 
> >>>>
> >>>
> >>>
> >>>True, and since some of those non-SIP features might not be 
> >>
> >>based on (IETF) standards, it might be good to have the 
> >>possibility for the developers of such applications to have a 
> >>short cut of saying what the app really is, rather than 
> >>having to list characteristics. In the end, some applications 
> >>are really not even meant to work with any other than the 
> >>exactly same application, for instance some networking games. 
> >>
> >>If the intent is to define a closed world, where 
> >>interoperability is not 
> >>even considered a desirable feature, then I agree. I think that may 
> >>indeed be the case for proprietary games.
> >>
> >>But for something like PTT, while it might be in the best 
> interest of 
> >>the maker of a particular PTT solution to keep it closed, I 
> >>don't think 
> >>it is in the best interest of society or the IETF that this 
> >>happen. This 
> >>is precisely the forum in which we try to prevent that from 
> happening.
> >>
> >>
> >>>>>Of course each of these proprietary services are allowed to 
> >>>>>define additional status or tuple-level extensions to PIDF.
> >>>>
> >>>>Yes, you can always extend the PIDF. My comment here is that 
> >>>>this might lead
> >>>>to an "explosion" of applications or services. It will 
> >>>>certainly not help
> >>>>interoperability.
> >>>
> >>My point precisely.
> >>
> >>
> >>>And who would standardize such characteristic definitions. 
> >>
> >>It has already taken a couple of years to standardize even 
> >>"prescaps", which is just a starting point. 
> >>
> >>>
> >>>>>However, I would like to have some kind of "service 
> >>>>>identifier" as part of the basic framework so that this could 
> >>>>>be done in a consistent manner. I think it would help 
> >>>>>especially in making of authorization and composition rules 
> >>>>>more simple. The current "class" attribute is way too 
> >>>>
> >>>>loosely defined.
> >>>>
> >>>>
> >>>>>So the concrete proposal is to include in RPID a "service id" 
> >>>>>element that would have a vendor-specific namespace, similar 
> >>>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
> >>>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
> >>>>><service-id>com.xyz.ptt</service-id>
> >>>>>while the PTT app compliant with OMA would have, e.g.
> >>>>><service-id>com.openmobilealliance.ptt</service-id>
> >>>>
> >>I am not fond of this solution. Done this way, we have defined an 
> >>element that has no semantics at all, only syntax.
> >>
> >>I would rather see vendor specific extensions to PIDF. At 
> >>least then the 
> >>people that support an extension will understand what it means.
> >>
> >>But I would rather see people come forward and do the hard work of 
> >>publicly defining new capabilities so that independent 
> >>implementations 
> >>can use them in an interoperable way.
> >>
> >>
> >>>>In general, I could agree with that approach. However, I 
> >>>>don't see a reason
> >>>>why the vendor specific SIP-based PTT should be different 
> >>>
> >>>>from the OMA based
> >>>
> >>>>PTT. Simply define one PTT, based on the SIP UA capabilities.
> >>>>
> >>>
> >>>
> >>>Well, for instance Nokia has a prorietary PTT application, 
> >>
> >>which uses SIP INVITE etc., so users are essentially 
> >>addressed with sip: URI. However, it has some floor control, 
> >>media etc. features which make it basically non-interoperable 
> >>with clients that are not specifically following that 
> >>proprietary spec. 
> >>
> >>This is a problem, not a good thing. I realize it has to 
> >>happen as part 
> >>of the development process, but the goal should be to get out 
> >>of it as 
> >>fast as possible.
> >>
> >>I would hope that endpoints supporting this would be 
> >>described as well 
> >>as possible using existing capabilities, and that a 
> >>reasonable effort be 
> >>made to do something reasonable when some other sip device connects.
> >>
> >>If you really can't deal with a regular voice caller at all, then I 
> >>would suggest, as a transitional step, saying that you 
> don't support 
> >>voice, and then advertising some proprietary capability as an 
> >>alternative to voice.
> >>
> >>
> >>>OMA PTT is closer to standard IETF SIP. Unless there is 
> >>
> >>some conversion, it does not work with Nokia PTT directly. 
> >>There should be some way for distinguishing these two in a 
> >>presence document, so that there won't be confusion. I think 
> >>the current idea is to do some kind of proprietary PIDF 
> >>extensions, but in my opinion a standard XML element in RPID 
> >>to tell this kind of information would be better, also for 
> >>the purposes of authorization rule making.
> >>
> >>See above for what I think you could do for now. The same 
> issues may 
> >>apply to OMA PTT, and a similar solution could be applied to it.
> >>
> >>Hopefully somebody will come up with a PTT that can be dealt 
> >>with more 
> >>gracefully. I think that work should be done on better defining 
> >>half-duplex, so that it can be used in conjunction with voice to 
> >>describe PTT. But it may just turn out to be the wrong thing. 
> >>Maybe we 
> >>need a "floor-control" capability.
> >>
> >>
> >>>>>in _addition_ to describing sip:, half-duplex audio, and the 
> >>>>>supported methods. Now:
> >>>>>1.) Watcher having one of those apps could see right away 
> >>>>>whether it is possible to communicate
> >>>>>2.) Composer getting PUBLISHes from 2 sources could 
> >>>>>immediately know that they are from a similar app
> >>>>>3.) The app could set its authorization rules using the 
> unique id.
> >>>>>
> >>>>>The main downside I can see in this approach that if there 
> >>>>>are two different proprietary apps that indeed could 
> >>>>>communicate at least partially, the watcher might make a 
> >>>>>conclusion that communication is not possible baed on 
> >>>>>different service-id.
> >>>>
> >>This is my concern as well.
> >>
> >> >>> However, in this case the watcher still
> >>
> >>>>>would have the characteristics visible and could determine 
> >>>>>that some interworking could be achieved. 
> >>>>
> >>>>Personally, I don't think it is a good idea that an end-user 
> >>>>should make
> >>>>such decission. If I get a presence document that indicates 
> >>>>that another
> >>>>person is able to receive some type of communication, I 
> >>>>expect that that
> >>>>type of communication would actually work when I use it.
> >>>
> >>With high probability. There are no guarantees with presence. For 
> >>instance see my mention above of codecs.
> >>
> >>
> >>>Exactly right. If you are an application who does not 
> >>
> >>necessarily aim for interoperability, but still uses sip:, 
> >>INVITE etc., you should be able to explicitely say so.
> >>
> >>And you can, without a special notion of service.
> >>
> >>
> >>>>>For this kind of 
> >>>>>reasons there should be careful text about when to use 
> >>>>>service-id in the first place, and what can be 
> determined from it.
> >>>>>
> >>>>>Does someone think that this is an absolutely bad idea? If 
> >>>>>so, how would you envision solving the issues discussed above?
> >>>>
> >>I still don't buy this.
> >>
> >>
> >>>>I think more care should be taken in mapping SIP UA 
> >>>>capabilities into actual
> >>>>services. That is not so clear to me!
> >>>
> >>>The idea is nice, but I don't see a problem in _addition_ 
> >>
> >>defining the application with something like 
> >>"com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind 
> >>of profile would have more meaning than the list of characteristics.
> >>
> >>I am still opposed to defining an attribute with no semantics.
> >>We have been around this so many times that I am quite 
> >>certain that it 
> >>will never be possible to get a useful agreement on what 
> >>"service" means.
> >>
> >>If it is introduced, I expect it to have problems like those 
> >>of the INFO 
> >>method. (The arguments for introducing it seem similar.)
> >>
> >>	Paul
> >>
> > 
> > 
> > _______________________________________________
> > Simple mailing list
> > Simple@ietf.org
> > https://www1.ietf.org/mailman/listinfo/simple
> > 
> 

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Subject: Re: [Simple] Presence Data Model: Identifying services
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inline.

Markus.Isomaki@nokia.com wrote:

> Hi Stefan,
> 
> Comments back to you inline.
> 
> (Yes, I absolutely agree with you on the usefulness of the Presence
> Data Model draft and that it should have been the starting point, at
> least right after PIDF.)

Thanks! Obviously I agree its a good thing ;)

> 
> To some extent that might be possible, but there might be differences
> e.g. in floor control etc., which make the service still practically
> unusable, which means that the presence document shouldn't give the
> impression that the communication will work. Obviously it is possible
> to list all those additional characteristics as well (and this might
> be a good idea too), but for instance if presence of a network game
> is published, it is easier to _explicitely_ tell what the game is by
> providing some sort of unique id which would be only meaningful to
> those similar game clients.

I think there are two aspects to this.

Firstly, I think we have a bunch of "SIP endpoints", like your PTT 
phone, which use SIP as a core technology, but don't interoperate with 
any other SIP endpoint. We've designed SIP to *be* interoperable, so if 
its not, its telling me that your protocol really isn't SIP after all. 
In such a case, I would suggest you register a new URI scheme for it, 
for example, ptt:user@domain, to indicate this. I think this is a 
scheme, since the point of a URI is to provide full context required to 
communicate. If sip:user@domain isn't providing that context since there 
are proprietary changes which prevent communication from compliant SIP 
endpoints, the scheme needs to be changed.

That said, there are cases where an endpoint is 100% sip compliant, but 
still it cannot communicate with another endpoint. Assuming proper 
behavior in terms of falling back to baseline usages when extensions 
aren't supported, I can only think of really a small number of reasons 
why this would happen:

1. Endpoints A and B do not support a common codec for the media types 
they have in common,

2. Endpoints A and B do not support a common set of media types


For the specific case of PTT, where "if you don't support this floor 
control technique, the call won't be set up", I put that in the first 
category (cases where its not really SIP). Thats because a SIP endpoint 
should be able to proceed with a call if extensions or features are not 
utilized. If you are not doing that because they are essential to the 
service, then it's something different.

The second case above (no common media or codecs) is solved in part in 
the presence document, where the supported media types are listed. We 
have elected not to include codecs in presence documents, so that case 
cannot be detected.

Of course, a call can always fail even if there are no protocol interop 
problems - I may call you, you insist on video, and since I don't do 
video, you elect not to proceed with the call. That's a **policy** 
problem, and I do not think it is the role of presence to generally 
convey the policies which are required to be in place for a call to 
complete. Perhaps others have different opinions.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: meaning of tel URI in presence data model, was: Re: [Simple] Presence
	Data Model: Identifying services
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inline.

Paul Kyzivat wrote:

> 
> 
> Aki Niemi wrote:
> 
>> On Thu, 2004-09-23 at 18:46, ext Paul Kyzivat wrote:
>>
>>> I don't find the scheme as clearcut as that. I don't think one should 
>>> assume "tel" indicates telephony. It perhaps does imply reachability 
>>> via various telephony protocols for purpose of voice communications, 
>>> but is not exclusive to that. If the tel: URI can be mapped to a sip 
>>> endpoint and protocol, then non-voice, non-telephony capabilities are 
>>> not excluded.
>>
>>
>> Hmm... If the scheme is that ambiguous, then perhaps its definition
>> needs to improve.
> 
> 
> Why? There is no particular reason why tel: should mean voice. We don't 
> restrict sip: that way, and http: doesn't say much about what you can do 
> either. (I can use http to exchange html, vxml, xcap, etc.)
> 
> AFAIK, tel: is an addressing format and namespace. The following is from 
>  draft-ietf-iptel-rfc2806bis-09:
> 
>    The "tel" URI telephone number is not restricted in the type of
>    termination point it refers to.  The termination point can be in the
>    public telephone network, a private telephone network or the
>    Internet. The termination point can be fixed or wireless and address
>    a fixed wired, mobile or nomadic terminal.  The terminal addressed
>    can support any electronic communication service (ECS) including
>    voice, data and fax.

Sigh... here is another case where the tel URI's confusion about what it 
really is, is getting us into trouble.

If you think about the tel URI as a urn scheme, whereby resolution to an 
actual set of communication resources is done using a resolution service 
(ala DDDS), then it becomes fairly obvious (to me at least) that you 
would simply never even put that URN into the <contact> of a presence 
document, since it doesnt represent a communications service at all. If 
anything, it's closer to a presentity identifier. Rather, you would put 
the individual service URIs that result from the DDDS lookups.

However, the tel URI is somewhat schizophrenic. Its not a URN, even 
though it can use (but doesnt require) URN-like resolution services such 
as enum. It's like a URN in that it can refer to an abstract name and be 
resolved to a wealth of different URIs of many different schemes and 
services. Its unlike a URN in that it is also used to refer to telephone 
resources, and the implication of the tel URI is that "I can dial this 
from a PSTN phone or enterprise phone and reach something". In that 
aspect, its more like a protocol URI.

What does this mean for presence? To be honest, in the discussion in the 
data model draft, I have assumed the protocol URI meaning. That is, it 
refers to a PSTN (or private network) circuit endpoint, period. That, of 
course, is a BIG assumption, and something we should discuss. I would 
propose that we keep this assumption and merely state it. In such a 
case, it would not make sense for the recipient of a presence document 
to look up  such a tel URI in enum. Rather, it knows its a telephony 
resource right away and can "dial it".

Comments?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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My guess is that Helsen is looking at something equivalent to the UAProf 
functions in WAP, which allow a client to push fairly complicated 
information about itself, such as screen size and JVM. I don't think 
this is "presence" in the sense that much of this information is 
probably not useful to watchers. However, it does represent valid device 
information and could conceivably appear there.

We do have a specification, RFC 3840, which allows a client to declare 
its capabilities to the server. That specification does not dictate how 
this information is used. One usage is caller preferences, RFC 3841, 
which allows for capability-based called routing. However, I think 
Helsen has in mind a usage where the provider/domain owner simply wants 
to look at this profile data for some reason, and that is certainly 
possible.

We have only a very limited set of capabilities declare right now, a 
much, much, much smaller subset of what is in UAProf.

-Jonathan R.



Paul Kyzivat wrote:

> Helsen,
> 
> I don't understand your terminology very well, or what you are trying to 
> achieve.
> 
> By 'client' I guess you mean a device addressed by a contact in a tuple 
> of the presence document, rather than a subscriber to presence. Is that 
> right?
> 
> I don't know precisely what role you have in mind for 'operator'. Is it 
> the operator of the presence server itself? Or is it something that is a 
> client of the PS. What is it doing that requires it to know this 
> information?
> 
> In general the device will probably be publishing its own presence 
> status to the presence server. If so the extension you suggest would 
> simply cause it to push this invariant information over and over.
> 
> This kind of info can already be polled using OPTIONS. Why isn't that 
> enough?
> 
>     Paul
> 
> Helsen Frank wrote:
> 
>> Hello,
>>
>>
>> At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a 
>> way to indicate to the server which client the user uses.  I can 
>> imagine that an operator has specific client capabilities he would 
>> like to know (e.g. java virtual machine version, screen size).  When 
>> including this client information, the server could retrieve the 
>> client capabilities from his e.g. his DB.  With this extension, the 
>> operator has control over the client information and is not dependend 
>> on the information the client provides to him.
>> Next example shows a possible extension
>>
>> <ClientInfo>
>>         <Manufacturor> Siemens </Manufacturor>
>>         <Model> CX65 </Model>
>>         <Version>2.103564</Version>
>> </ClientInfo>
>>
>> Regards,
>>
>> Frank
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
> 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Simple] Presence Data Model: Identifying services
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To: ext Paul Kyzivat <pkyzivat@cisco.com>
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Inline.

On Fri, 2004-09-24 at 16:59, ext Paul Kyzivat wrote: 
> > Hmm... If the scheme is that ambiguous, then perhaps its definition
> > needs to improve.
> 
> Why? There is no particular reason why tel: should mean voice. We don't 
> restrict sip: that way, and http: doesn't say much about what you can do 
> either. (I can use http to exchange html, vxml, xcap, etc.)

There is a big difference. I can safely assume that for sip:, I am able
to send a SIP request and the other end will understand it; but probably
not an HTTP request, or let's say an SMTP message using that URI.

I admit tel is more problematic, since it is more of a name, really
without a protocol attached to it, and sending a SIP request to it might
actually work (because we've defined what that means). But the inverse
isn't true, since POTS phones don't do IM or anything fancy like that.
That's why I think it's futile to build support for them on top of tel.

BTW, there is also really no way to know whether I can send an SMS to
that telephone number (outside of recognizing it as a mobile number).
Which is one of the reasons an explicit smsto URI would be nicer...

> AFAIK, tel: is an addressing format and namespace. The following is from 
>   draft-ietf-iptel-rfc2806bis-09:
> 
>     The "tel" URI telephone number is not restricted in the type of
>     termination point it refers to.  The termination point can be in the
>     public telephone network, a private telephone network or the
>     Internet. The termination point can be fixed or wireless and address
>     a fixed wired, mobile or nomadic terminal.  The terminal addressed
>     can support any electronic communication service (ECS) including
>     voice, data and fax.

Right, but I definitely don't think that IM or voice+IM session are
ECSs. SIP happens to have a linkage in that if I INVITE a tel URI with
appropriate media (e.g., voice or fax), then it may work if there is an
appropriate GW in place or a termination point that speaks SIP.

> >>For instance, it may prove convenient to publish a tel URI for a sip 
> >>endpoint that is capable of both voice and IM. Then calls via the
> PSTN 
> >>will work, calls from a sip audio phone will work, calls from
> another 
> >>voice+im client will have access to voice and IM.
> > 
> > IMO the most obvious solution here is that if the endpoint wants to
> > advertize services other than (or in addition to) telephony, it
> > publishes a SIP or IM tuple as well as a tel tuple. Is there a
> reason
> > this would not work?
> 
> I don't see a problem here, and so I don't feel a need for a solution.
> 
> I agree that other address forms are probably desirable in this case, 
> for a variety of reasons. But that doesn't mean we should
> automatically 
> assume tel: implies voice telephony and only that.

I think we have to assume tel implies some telephony, or ECS service
(whatever that means). If a presentity wants to advertize services
beyond those, then it is only reasonable that those other contacts are
also shown in presence.

> One place where this situation may well come up is when the tel: uri
> is 
> used as the presentity address. (Something that should be ok, and
> makes 
> sense to use if you are a client and that is all you have.) It may be 
> that the user doesn't want to expose device addresses, and so provides
> only the presentity address back as the contact in the tuple(s).

I don't understand what the problem there is. Even if tel URI was
allowed as the entity attribute in PIDF (which it isn't AFAIK) then that
presence document would simply state the status of telephony services,
nothing more.

Cheers,
Aki



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All,

It might not be in the best interest of the user to broadcast too much 
detail about the configuration of the device as this might just let 
miscreants know what sort of tools to use to attack the device.  May save 
some reconaissance/scanning time on the attacker's part.

Mike

At 11:47 AM 9/24/2004 -0400, Jonathan Rosenberg wrote:
>My guess is that Helsen is looking at something equivalent to the UAProf 
>functions in WAP, which allow a client to push fairly complicated 
>information about itself, such as screen size and JVM. I don't think this 
>is "presence" in the sense that much of this information is probably not 
>useful to watchers. However, it does represent valid device information 
>and could conceivably appear there.
>
>We do have a specification, RFC 3840, which allows a client to declare its 
>capabilities to the server. That specification does not dictate how this 
>information is used. One usage is caller preferences, RFC 3841, which 
>allows for capability-based called routing. However, I think Helsen has in 
>mind a usage where the provider/domain owner simply wants to look at this 
>profile data for some reason, and that is certainly possible.
>
>We have only a very limited set of capabilities declare right now, a much, 
>much, much smaller subset of what is in UAProf.
>
>-Jonathan R.
>
>
>
>Paul Kyzivat wrote:
>
>>Helsen,
>>I don't understand your terminology very well, or what you are trying to 
>>achieve.
>>By 'client' I guess you mean a device addressed by a contact in a tuple 
>>of the presence document, rather than a subscriber to presence. Is that right?
>>I don't know precisely what role you have in mind for 'operator'. Is it 
>>the operator of the presence server itself? Or is it something that is a 
>>client of the PS. What is it doing that requires it to know this information?
>>In general the device will probably be publishing its own presence status 
>>to the presence server. If so the extension you suggest would simply 
>>cause it to push this invariant information over and over.
>>This kind of info can already be polled using OPTIONS. Why isn't that enough?
>>     Paul
>>Helsen Frank wrote:
>>
>>>Hello,
>>>
>>>
>>>At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a way 
>>>to indicate to the server which client the user uses.  I can imagine 
>>>that an operator has specific client capabilities he would like to know 
>>>(e.g. java virtual machine version, screen size).  When including this 
>>>client information, the server could retrieve the client capabilities 
>>>from his e.g. his DB.  With this extension, the operator has control 
>>>over the client information and is not dependend on the information the 
>>>client provides to him.
>>>Next example shows a possible extension
>>>
>>><ClientInfo>
>>>         <Manufacturor> Siemens </Manufacturor>
>>>         <Model> CX65 </Model>
>>>         <Version>2.103564</Version>
>>></ClientInfo>
>>>
>>>Regards,
>>>
>>>Frank
>>>
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>Simple mailing list
>>>Simple@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/simple
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>_______________________________________________
>Simple mailing list
>Simple@ietf.org
>https://www1.ietf.org/mailman/listinfo/simple


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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - Chunking
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Out approach to chunking has evolved thru the versions, and the document
doesn't seem wholely clear and consistent about what we have arrived at.
In fact, I am not entirely certain I know fully what we have decided,
and I can't figure it out for certain from the document. I would propose
specific wording changes, but first I want to get clarification about
how we think it should now work.

Based on scattered information in the doc, and reading between the
lines, and factoring in stuff from conversations along the way, I
*think* the following is intended:

- only SEND messages are chunkable

- there are two kinds of chunks: interrruptible and uninterrupible.

- a chunk is uninterruptible if it has a Byte-Range header with a
    numeric range-end.

- other chunks are interruptible. (No Byte-Range, or a Byte-Range
    with a range-end of "*".)

- uninterruptible chunks MUST NOT have a range-end greater than 2048.

- chunks other than the last SHOULD (or MUST?) have a body ('data'
    in the syntax) of length exactly 2048. If SHOULD, what is a
    valid reason to violate?

- responses never have bodies at all. Requests other than SEND
    may have a body but are unchunkable. The presence of a Byte-Range
    in a request other than SEND says nothing about the body of that
    request. (In REPORT it refers to the SEND being reported on.
    See separate thread on Byte-Range.) This at best seems irregular
    and weird.

Section 6.3.1 says: "it is possible the body is shorter than the
range-end field indicates. This can occur if the sender interrupted a
SEND request unexpectedly." Based on what I have interpretted above,
this is no longer possible. (To be interrupted it must have had a
range-end of "*".)



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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - nits
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I just did a fairly careful review of
draft-ietf-simple-message-sessions-08. I came up with a bunch of comments.

I will include the nits here, but in the interest of discussion I will
break the rest up into a number of separate threads.

	Paul

Section 3, near the end: "This document specifies MSRP behavior only
peer-to-peer session...". Is awkward wording. Suggest inserting "for a"
in front of "peer-to-peer".

Section 4.3, last paragraph: "... but some other systems choose to use a
value of "partial" to reduce the load on the servers caused by 200 OK
responses, but still allow error responses to be sent in many cases."
This will make sense later, but it doesn't make sense at this point in
the document because the linkage between the report and the 200 hasn't
been made. Suggest something like:

"... but some other systems choose to use a value of "partial" to reduce
the load on the servers. As will be seen in section 6.2, this will omit
200 OK responses and the load they cause, but still allow error
responses to be sent in many cases."

Section 4.4, 4th paragraph from end, there is a sentence fragment that
should be combined with the next sentence:
"If the receiving endpoint receives more than one message with the same
Message-ID. It SHOULD assume that the messages are duplicates."
Should be:
"If the receiving endpoint receives more than one message with the same
Message-ID it SHOULD assume that the messages are duplicates."

The word "insure" is used twice in this document, and in both cases is
should be "ensure".

Section 6.1.2 says: "An MSRP endpoint MUST be able to generate success
REPORT requests." But nowhere do I find a comparable statement about
failure reports. I think the endpoint MUST be able to do both.

Section 6.1.3, 1st paragraph: there is an instance of "respons" that
should be "response".

Section 6.3.1: s/If chunks data/If chunk data/

Section 7.1 says to use "*" to indicate willingness to accept any
content type. This has always seemed gratuitous to me, since the mime
syntax for types already allows "*/*" to mean that. I think it would
minimize future issues if we just accepted that instead of specifying
"*" as a special case. This would require a small change to the syntax 
of 'type' and 'subtype'.

Section 7.1 uses both "container types" and "compound types" to mean the
same thing. Can we just pick one and stick with it? (I don't care which
one.)

Section 7.1.5 says: "Instead, MSRP now specifies a default connection
direction." Its not a default, since you can't change it. I think what
is meant is: "MSRP now associates connection direction with the
direction the offer travels."

Section 10.5: Shows using a REGISTER to query for the contacts. There
really isn't anything wrong here, but it may cause some readers to
conclude that using REGISTER this way is something that should normally
be done. Might be worth a disclaimer that this is only shown for
purposes of clarifying the example.




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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
	Overlapping Chunks
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Section 6.3.1 discusses how to handle the receipt of overlapping chunks,
specifying that the last received copy of any particular byte wins. I
don't think this is good.

If the overlapping chunks contain consistent data then it doesn't matter
which takes precedence. If they contain conflicting data then reordering
of chunks can affect the result unpredictably. This rule also prevents
the recipient from rendering a byte when it and all preceding bytes have
been received.

I think it should be permissible for the recipient to check for
inconsistency in the overlapping case, and report an error if so. We may
want to require this checking. (Since it typically won't happen it
shouldn't have a performance impact.)



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From simple-bounces@ietf.org  Fri Sep 24 12:48:13 2004
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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - Byte
 Ranges in REPORTs
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I am really confused about how byte ranges are supposed to work in
reports, and I suspect we are permitting a bunch of nonsense.

Fundamentally, a message can only be considered a success if all its
bytes are received successfully. If it is to be positively acknowledged
   with success reports, then there must be success reports covering
every byte. Is there any point in sending a success report for less than
the entire message? It doesn't help the receiver, and actually makes
more work because it then requires keeping track of what bytes have been
confirmed. And it doesn't ease the job of the sender, who could easily
just wait until all has been received before sending the success report.

The flip side is that a message fails if a single byte fails. At that
point it does no good to know that other parts succeeded or failed. So
while it makes sense to send a failure report based on problems with a
single chunk, there is little value in saying which byte range failed.

Bottom line: unless I have missed something, there is no reason for
Byte-Range in REPORT or responses. A success report should only be sent
after all the chunks of a message have been received. A failure report,
or an error status, can be sent at the time the failure of a single
chunk is known, but need not identify the byte range. (In that case,
maybe the Byte-Range could be optional for debugging purposes only.)




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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - Port in
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Section 7.1 says: "An offered or accepted MSRP media-line MUST have the
following value exactly ... m=message 9 msrp *"

We've been thru this before. It can't always be 9 because there could be
two media sections and they can't both use 9. I think we say it must be:

    m=message <port> msrp *

and then say <port> SHOULD be 9, but MAY be something else if 9 has
already been used.




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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - Syntax
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Section 5: MSRP URL syntax:

There is an undefined reference to "ALPHANUM". RFC2396 defines 
"alphanum", but that is only a single byte. Suggest replacing ALPHANUM 
with 1*alphanum, and reference 2396 for alphanum.

Rohan wanted "?" syntax added.

Section 8: Formal Syntax:

I don't know what resulted in the following:

     dUmMy= "Status:" SP namespace SP short-status [SP text-reason]

It must have been intentional. But the 'Status' element is undefined, so 
I think this needs to be changed to:

     Status = "Status:" SP namespace SP short-status [SP text-reason]

We define pval as:

     pval = token / quoted-string

We might want to consider making it compatible with generic-param in 
3261, which also permits it to be a 'host'.



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Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 - max-size
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Section 7.1 says: "An endpoint MAY indicate the maximim size message
they wish to receive using the max-size a-line attribute".

Does this apply only to a *message* - namely the content of a SEND
request? Or does this apply to any and all requests (including future
extensions) that have content?




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Don't forget to add 405 for reporting a method that isn't understood.

Example Bugs:

Section 10.1, step 4: CRLF missing before the "Hi"

Section 10.1, steps 5 & 7: To-Path and From-Path values are backward.




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Subject: Re: meaning of tel URI in presence data model, was: Re:
	[Simple] Presence Data Model: Identifying services
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At 11:37 AM 9/24/2004 -0400, Jonathan Rosenberg wrote:
>inline.
>
>Paul Kyzivat wrote:
>
>>
>>Aki Niemi wrote:
>>
>>>On Thu, 2004-09-23 at 18:46, ext Paul Kyzivat wrote:
>>>
>>>>I don't find the scheme as clearcut as that. I don't think one should 
>>>>assume "tel" indicates telephony. It perhaps does imply reachability 
>>>>via various telephony protocols for purpose of voice communications, 
>>>>but is not exclusive to that. If the tel: URI can be mapped to a sip 
>>>>endpoint and protocol, then non-voice, non-telephony capabilities are 
>>>>not excluded.
>>>
>>>
>>>Hmm... If the scheme is that ambiguous, then perhaps its definition
>>>needs to improve.
>>
>>Why? There is no particular reason why tel: should mean voice. We don't 
>>restrict sip: that way, and http: doesn't say much about what you can do 
>>either. (I can use http to exchange html, vxml, xcap, etc.)
>>AFAIK, tel: is an addressing format and namespace. The following is 
>>from  draft-ietf-iptel-rfc2806bis-09:
>>    The "tel" URI telephone number is not restricted in the type of
>>    termination point it refers to.  The termination point can be in the
>>    public telephone network, a private telephone network or the
>>    Internet. The termination point can be fixed or wireless and address
>>    a fixed wired, mobile or nomadic terminal.  The terminal addressed
>>    can support any electronic communication service (ECS) including
>>    voice, data and fax.
>
>Sigh... here is another case where the tel URI's confusion about what it 
>really is, is getting us into trouble.
>
>If you think about the tel URI as a urn scheme, whereby resolution to an 
>actual set of communication resources is done using a resolution service 
>(ala DDDS), then it becomes fairly obvious (to me at least) that you would 
>simply never even put that URN into the <contact> of a presence document, 
>since it doesnt represent a communications service at all. If anything, 
>it's closer to a presentity identifier. Rather, you would put the 
>individual service URIs that result from the DDDS lookups.
>
>However, the tel URI is somewhat schizophrenic. Its not a URN, even though 
>it can use (but doesnt require) URN-like resolution services such as enum. 
>It's like a URN in that it can refer to an abstract name and be resolved 
>to a wealth of different URIs of many different schemes and services. Its 
>unlike a URN in that it is also used to refer to telephone resources, and 
>the implication of the tel URI is that "I can dial this from a PSTN phone 
>or enterprise phone and reach something". In that aspect, its more like a 
>protocol URI.
>
>What does this mean for presence? To be honest, in the discussion in the 
>data model draft, I have assumed the protocol URI meaning. That is, it 
>refers to a PSTN (or private network) circuit endpoint, period. That, of 
>course, is a BIG assumption, and something we should discuss. I would 
>propose that we keep this assumption and merely state it. In such a case, 
>it would not make sense for the recipient of a presence document to look 
>up  such a tel URI in enum. Rather, it knows its a telephony resource 
>right away and can "dial it".
>
>Comments?

There was some discussion on one of the mail lists that a parameter was 
needed to indicate that an ENUM dip had already been performed and 
determined that this particular terminal must be reached by going to the 
PSTN, i.e., that it was a circuit endpoint.  That could be one of the 
things that the presence document indicates.

If one were to do an ENUM lookup and receive a list of indications of IP 
endpoints, well, I would have expected that list to have been included in 
the presence for that user in the first place.

I think it much more useful for ENUM to point to a presence address leading 
to richer data with corresponding user controls than the reverse.  So, I 
guess I'm in agreement with Jonathan.

Mike



>-Jonathan R.
>
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
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Goeman Stefan wrote:
> Hello,
> 
> Comments inline.
> 
> Greetings,
> Stefan.
> 
> 
>>>On second thought, I realized that there is one problem:
>>>How will this work together with the Authorization rules.
>>
>>I think you are imagining a problem here. I don't think it makes much 
>>sense for the client to determine authorization, beyond simply being 
>>authorized to subscribe to presence. If the PS gives you the 
>>info, just 
>>display it. The PS can filter out what it thinks you should 
>>not see or do.
>>
> 
> I think we are talking about different things here. I was taking about me as
> presentity managing and editing my own authorization rules. Since my
> presence information is also "my property" I would expect that I would be
> able to play the role of a rule maker. Clearly, you have to edit your rules
> somehow, via some device.

Oh, sorry. Yes, the presentity does have to manage rules for what will 
be provided to subscribers. There is already much work on that.

	Paul

>>In any case there is no good way for a client to determine if it is 
>>permitted to do something, short of trying to do it.
>>
>>	Paul
>>
>> > Here, again, you
>>
>>>have to present the authorization rules in a meaningful 
>>
>>manner (via some
>>
>>>front end user interface). If I am correct, the idea is to 
>>
>>retrieve the
>>
>>>authorization rules via XCAP from the policy server, make 
>>
>>some changes and
>>
>>>provide the changes back to the policy server via XCAP.  
>>
>>Yet, the policy
>>
>>>server is not aware of any services like IM, voice call, 
>>
>>... . So, again,
>>
>>>the device should make this mapping. This works well if you 
>>
>>only use one
>>
>>>device. But, if you use more than one device, with 
>>
>>different capabilities
>>
>>>and services, it does not work anymore. A device can't say something
>>>meaningful over a service it does not support. 
>>
>>Consequently, you can not
>>
>>>edit every aspect of the authorization rules on your 
>>
>>device. I think this is
>>
>>>problematic, and I don't see a good solution right know.
>>>
>>>Greetings,
>>>Stefan. 
>>>
>>>
>>>
>>>>-----Original Message-----
>>>>From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>>>>Sent: donderdag 23 september 2004 17:47
>>>>To: Markus.Isomaki@nokia.com
>>>>Cc: Goeman Stefan; simple@ietf.org
>>>>Subject: Re: [Simple] Presence Data Model: Identifying services
>>>>
>>>>
>>>>I have comments both to Markus' original posting and to the ensuing 
>>>>discussion. So I will just insert inline in the latest 
>>>>response I have.
>>>>
>>>>	Paul
>>>>
>>>>Markus.Isomaki@nokia.com wrote:
>>>>
>>>>
>>>>>Hi Stefan,
>>>>>
>>>>>Comments back to you inline. 
>>>>>
>>>>>(Yes, I absolutely agree with you on the usefulness of the 
>>>>
>>>>Presence Data Model draft and that it should have been the 
>>>>starting point, at least right after PIDF.)
>>>>
>>>>
>>>>>>-----Original Message-----
>>>>>>From: simple-bounces@ietf.org 
>>>>>>[mailto:simple-bounces@ietf.org]On Behalf
>>>>>>Of ext Goeman Stefan
>>>>>>Sent: 23 September, 2004 14:42
>>>>>>To: 'simple@ietf.org'
>>>>>>Subject: RE: [Simple] Presence Data Model: Identifying services
>>>>>>
>>>>>>
>>>>>>Hello,
>>>>>>
>>>>>>Comments inline
>>>>>>
>>>>>>P.S.: In general, I find the Presence Data model draft a good 
>>>>>>document. This
>>>>>>is how things should have started in the first place.
>>>>>>
>>>>>>Greetings,
>>>>>>Stefan.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>>-----Original Message-----
>>>>>>>From: simple-bounces@ietf.org 
>>>>>>>[mailto:simple-bounces@ietf.org] On Behalf Of 
>>>>>>
>>>>>>Markus.Isomaki@nokia.com
>>>>>>
>>>>>>
>>>>>>
>>>>>>>Sent: donderdag 23 september 2004 10:10
>>>>>>>To: simple@ietf.org
>>>>>>>Subject: [Simple] Presence Data Model: Identifying services
>>>>>>>
>>>>>>>
>>>>>>>Hi all,
>>>>>>>
>>>>>>>I have a comment on Presence Data Model draft about the 
>>>>>>>identification/characterization of services. (I know this has 
>>>>>>>been discussed for a while already, but I still think the 
>>>>>>>current way is somewhat unadequate.)
>>>>>>>
>>>>>>>The current Data Model draft says:
>>>>>>>
>>>>>>> In this model, services are not explicitly enumerated. 
>>>>>>
>> That is,
>>
>>>>>>> there is no "service" attribute with values of "ptt" or 
>>>>>>>"telephony".
>>>>>>> Rather, the service is identified in one of two ways.  In 
>>>>>>>many cases,
>>>>>>> the URI scheme is a clear indicator of service.  An "sms" 
>>>>>>>URI clearly
>>>>>>> indicates SMS, and a "tel" URI clearly indicates 
>>>>>>>telephony. 
>>>>>>
>>>>I don't find the scheme as clearcut as that. I don't think 
>>>
>>one should 
>>
>>>>assume "tel" indicates telephony. It perhaps does imply 
>>>>reachability via 
>>>>various telephony protocols for purpose of voice 
>>>>communications, but is 
>>>>not exclusive to that. If the tel: URI can be mapped to a sip 
>>>>endpoint 
>>>>and protocol, then non-voice, non-telephony capabilities are 
>>>>not excluded.
>>>>
>>>>For instance, it may prove convenient to publish a tel URI 
>>>
>>for a sip 
>>
>>>>endpoint that is capable of both voice and IM. Then calls via 
>>>>the PSTN 
>>>>will work, calls from a sip audio phone will work, calls 
>>>
>>from another 
>>
>>>>voice+im client will have access to voice and IM.
>>>>
>>>>
>>>>>>>For some
>>>>>>
>>>>>>> URIs, there may be many services available, for example, 
>>>>>>
>>>>>>SIP.  For
>>>>>>
>>>>>>
>>>>>>
>>>>>>> those services, each service has a set of 
>>>>>>
>>>>>>characteristics, each of
>>>>>>
>>>>>>
>>>>>>
>>>>>>> which has a well-defined meaning, such that a system can
>>>>>>> unequivocally determine whether or not the service has that
>>>>>>> characteristic.  This is discussed in more detail in [5]. 
>>>>>>>
>>>>>>>I agree that the contact URI scheme is the most important 
>>>>>>>piece of information to distinguish what is the "service". 
>>>>>>>(There are some gaps here too, since e.g. the URI schemes for 
>>>>>>>SMS or MMS do not even exist, and there may be other non-IETF 
>>>>>>>protocols that face the same problem.) However, I'm not 
>>>>>>>convinced that listing the signaling/media characteristics of 
>>>>>>>the end-point or service really gives enough information to 
>>>>>>>the watcher to really determine if it can communicate with 
>>>>>>>the advertised service/application. 
>>>>>>>
>>>>>>>The refenced draft [5] has a following example:
>>>>>>>
>>>>>>>5.6 Walkie-talkie
>>>>>>>
>>>>>>> The walkie-talkie service allows real-time voice communication
>>>>>>> between participants.  Only one participant can speak at a 
>>>>>>>time; that
>>>>>>> is, communication is half-duplex.  Typically, 
>>>>>>
>>>>>>participants press a
>>>>>>
>>>>>>
>>>>>>
>>>>>>> button to indicate that they are ready to speak, although other
>>>>>>> mechanism (e.g.  voice activation) are occasionally used.
>>>>>>>
>>>>>>> Support for the Walkie-Talkie service can be deduced by 
>>>>>>>observing the
>>>>>>> presence of a contact URI with a scheme of "sip:", 
>>>>>>
>>>>>>associated with
>>>>>>
>>>>>>
>>>>>>
>>>>>>> the following minimal set of capabilities: <audio>true</audio>
>>>>>>> <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>>>>>>
>>>>>>>Presumably this is the same as the service sometimes called 
>>>>>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
>>>>>>>are several _non-interworking_ variants of this service, that 
>>>>>>>still all would match the characteristics listed above, as 
>>>>>>>the _SIP_ part might still be standard, at least for some 
>>>>>>>methods etc. But still the communication would fail even if 
>>>>>>>the tuple looks OK.
>>>>>>
>>>>>>I don't think I understand you here. I would expect that if 
>>>>>>both end-point
>>>>>>SIP US support exactly the same methods, the SIP UA should 
>>>>>>also be able to
>>>>>>handle those SIP method. Or, to phase it in another way, both 
>>>>>>end-point
>>>>>>should be able to communicate with each other only based on 
>>>>>>using these
>>>>>>capabilities.
>>>>>>
>>>>>
>>>>>
>>>>>To some extent that might be possible, but there might be 
>>>>
>>>>differences e.g. in floor control etc., which make the 
>>>>service still practically unusable, which means that the 
>>>>presence document shouldn't give the impression that the 
>>>>communication will work. Obviously it is possible to list all 
>>>>those additional characteristics as well (and this might be a 
>>>>good idea too), but for instance if presence of a network 
>>>>game is published, it is easier to _explicitely_ tell what 
>>>>the game is by providing some sort of unique id which would 
>>>>be only meaningful to those similar game clients.
>>>>
>>>>>>>(Also if I saw the characteristics above, 
>>>>>>>I could as well determine that they describe normal VoIP 
>>>>>>>application running in (very) some old PC with half-duplex 
>>>>>>>audio drivers, so I claim the service description is hard to 
>>>>>>>make unambiguous in the first place.)  
>>>>>>
>>>>I think the goal should be that all of these are at least minimally 
>>>>interoperable. I ought to be able to call a PTT contact from 
>>>>a regular 
>>>>voice phone and have *something* reasonable happen. (Maybe 
>>>
>>it is just 
>>
>>>>oneway voice, with the non-ptt caller never getting the 
>>>>floor, but that 
>>>>may be better than nothing.
>>>>
>>>>I think the real problem in this case is not with the capability 
>>>>approach. It is that the 'duplex' capability is 
>>>>underspecified. There is 
>>>>no explanation of how a pair of half duplex endpoints (or one 
>>>>half-duplex and one full-duplex endpoint) agree on who can talk.
>>>>
>>>>In the case of the 'voice' capability, it is understood that 
>>>>having both 
>>>>endpoints support voice isn't enough to support 
>>>>communication. There is 
>>>>an SDP negotiation of codecs, etc. that either succeeds or 
>>>>fails. There 
>>>>should be something similar for half-duplex.
>>>>
>>>>
>>>>
>>>>>>I also have the impression that it will be difficult to 
>>>>>
>>>>determine the
>>>>
>>>>
>>>>>>services merely based on the SIP methods (and ...) supported 
>>>>>>by the SIP UA.
>>>>>>It possibly might mean that the SIP-based ptt-service on my 
>>>>>>mobile phone
>>>>>>might be mapped onto some communication service on your PC. 
>>>>>>Then, I wonder
>>>>>>who will be responsible for such kind of mapping? Only the 
>>>>>>end terminals, or
>>>>>>is support from the network, presence server, necessary? And, 
>>>>>>how will one
>>>>>>service in one domain be mapped into a service in another 
>>>>>>domain? I believe
>>>>>>there are some open issues here. 
>>>>>
>>>>The intent is that capabilities have common meanings in all 
>>>>domains, so 
>>>>they don't need to be mapped. I don't see why there should be any 
>>>>binding between services and domains.
>>>>
>>>>In the general case one should assume that there are devices 
>>>>of widely 
>>>>varying capabilities in every domain, and that they know best 
>>>>what they 
>>>>are capable of. So the device should be the one deciding if 
>>>
>>a contact 
>>
>>>>with and advertised set of capabilties is something it wants 
>>>>to attempt 
>>>>communication with.
>>>>
>>>>In some environment where there is an intermediary in front 
>>>>of a device 
>>>>that understands the device and acts on its behalf, then I 
>>>
>>suppose it 
>>
>>>>could make these decisions. But I would consider that the odd case.
>>>>
>>>>
>>>>
>>>>>I think that it is fully OK to describe some rather simple 
>>>>
>>>>things using characteristics, but in many cases applications 
>>>>are more complex.
>>>>
>>>>Clearly we have chosen not to describe capabilities if full detail. 
>>>>(E.g. we don't publish codec support.) Because of this, 
>>>
>>there is the 
>>
>>>>possibility that we make a wrong decision about ability to 
>>>>communicate 
>>>>and it doesn't get discovered until we try.
>>>>
>>>>This is ok when the expected probability of failure is low. (We 
>>>>generally expect that there will be some codec in common, 
>>>
>>or that one 
>>
>>>>end or the other will be able to introduce a transcoder.)
>>>>
>>>>
>>>>
>>>>>>>I think the main point of a service tuple is to give the 
>>>>>>>watcher enough information to know whether he can really 
>>>>>>>communicate & interoperate with the advertised service. Given 
>>>>>>>that many services taht are using SIP also have propritary or 
>>>>>>>non-SIP features, I don't think the current approach is enough. 
>>>>>>
>>>>In cases where the existing capabilities aren't sufficient to 
>>>>ensure a 
>>>>good probability of communication, then we should indeed 
>>>>consider adding 
>>>>additional capabilities.
>>>>
>>>>But I continue to object to the notion of "service" as the 
>>>
>>way to do 
>>
>>>>that, because it conveys no semantics about what the 
>>>>capability is that 
>>>>it describes.
>>>>
>>>>
>>>>
>>>>>>Do you have an example of a service that uses SIP and also has non
>>>>>>SIP-features? If your service also uses non-SIP features, 
>>>>>
>>>>you can not
>>>>
>>>>
>>>>>>determine the type of service by only looking at the SIP 
>>>>>
>>>>capabilities
>>>>
>>>>
>>>>>>supported by the SIP UA. Some indication of that non-SIP 
>>>>>>feature should also
>>>>>>be present in the presence document. 
>>>>>>
>>>>>
>>>>>
>>>>>True, and since some of those non-SIP features might not be 
>>>>
>>>>based on (IETF) standards, it might be good to have the 
>>>>possibility for the developers of such applications to have a 
>>>>short cut of saying what the app really is, rather than 
>>>>having to list characteristics. In the end, some applications 
>>>>are really not even meant to work with any other than the 
>>>>exactly same application, for instance some networking games. 
>>>>
>>>>If the intent is to define a closed world, where 
>>>>interoperability is not 
>>>>even considered a desirable feature, then I agree. I think that may 
>>>>indeed be the case for proprietary games.
>>>>
>>>>But for something like PTT, while it might be in the best 
>>>
>>interest of 
>>
>>>>the maker of a particular PTT solution to keep it closed, I 
>>>>don't think 
>>>>it is in the best interest of society or the IETF that this 
>>>>happen. This 
>>>>is precisely the forum in which we try to prevent that from 
>>>
>>happening.
>>
>>>>
>>>>>>>Of course each of these proprietary services are allowed to 
>>>>>>>define additional status or tuple-level extensions to PIDF.
>>>>>>
>>>>>>Yes, you can always extend the PIDF. My comment here is that 
>>>>>>this might lead
>>>>>>to an "explosion" of applications or services. It will 
>>>>>>certainly not help
>>>>>>interoperability.
>>>>>
>>>>My point precisely.
>>>>
>>>>
>>>>
>>>>>And who would standardize such characteristic definitions. 
>>>>
>>>>It has already taken a couple of years to standardize even 
>>>>"prescaps", which is just a starting point. 
>>>>
>>>>
>>>>>>>However, I would like to have some kind of "service 
>>>>>>>identifier" as part of the basic framework so that this could 
>>>>>>>be done in a consistent manner. I think it would help 
>>>>>>>especially in making of authorization and composition rules 
>>>>>>>more simple. The current "class" attribute is way too 
>>>>>>
>>>>>>loosely defined.
>>>>>>
>>>>>>
>>>>>>
>>>>>>>So the concrete proposal is to include in RPID a "service id" 
>>>>>>>element that would have a vendor-specific namespace, similar 
>>>>>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
>>>>>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
>>>>>>><service-id>com.xyz.ptt</service-id>
>>>>>>>while the PTT app compliant with OMA would have, e.g.
>>>>>>><service-id>com.openmobilealliance.ptt</service-id>
>>>>>>
>>>>I am not fond of this solution. Done this way, we have defined an 
>>>>element that has no semantics at all, only syntax.
>>>>
>>>>I would rather see vendor specific extensions to PIDF. At 
>>>>least then the 
>>>>people that support an extension will understand what it means.
>>>>
>>>>But I would rather see people come forward and do the hard work of 
>>>>publicly defining new capabilities so that independent 
>>>>implementations 
>>>>can use them in an interoperable way.
>>>>
>>>>
>>>>
>>>>>>In general, I could agree with that approach. However, I 
>>>>>>don't see a reason
>>>>>>why the vendor specific SIP-based PTT should be different 
>>>>>
>>>>>>from the OMA based
>>>>>
>>>>>
>>>>>>PTT. Simply define one PTT, based on the SIP UA capabilities.
>>>>>>
>>>>>
>>>>>
>>>>>Well, for instance Nokia has a prorietary PTT application, 
>>>>
>>>>which uses SIP INVITE etc., so users are essentially 
>>>>addressed with sip: URI. However, it has some floor control, 
>>>>media etc. features which make it basically non-interoperable 
>>>>with clients that are not specifically following that 
>>>>proprietary spec. 
>>>>
>>>>This is a problem, not a good thing. I realize it has to 
>>>>happen as part 
>>>>of the development process, but the goal should be to get out 
>>>>of it as 
>>>>fast as possible.
>>>>
>>>>I would hope that endpoints supporting this would be 
>>>>described as well 
>>>>as possible using existing capabilities, and that a 
>>>>reasonable effort be 
>>>>made to do something reasonable when some other sip device connects.
>>>>
>>>>If you really can't deal with a regular voice caller at all, then I 
>>>>would suggest, as a transitional step, saying that you 
>>>
>>don't support 
>>
>>>>voice, and then advertising some proprietary capability as an 
>>>>alternative to voice.
>>>>
>>>>
>>>>
>>>>>OMA PTT is closer to standard IETF SIP. Unless there is 
>>>>
>>>>some conversion, it does not work with Nokia PTT directly. 
>>>>There should be some way for distinguishing these two in a 
>>>>presence document, so that there won't be confusion. I think 
>>>>the current idea is to do some kind of proprietary PIDF 
>>>>extensions, but in my opinion a standard XML element in RPID 
>>>>to tell this kind of information would be better, also for 
>>>>the purposes of authorization rule making.
>>>>
>>>>See above for what I think you could do for now. The same 
>>>
>>issues may 
>>
>>>>apply to OMA PTT, and a similar solution could be applied to it.
>>>>
>>>>Hopefully somebody will come up with a PTT that can be dealt 
>>>>with more 
>>>>gracefully. I think that work should be done on better defining 
>>>>half-duplex, so that it can be used in conjunction with voice to 
>>>>describe PTT. But it may just turn out to be the wrong thing. 
>>>>Maybe we 
>>>>need a "floor-control" capability.
>>>>
>>>>
>>>>
>>>>>>>in _addition_ to describing sip:, half-duplex audio, and the 
>>>>>>>supported methods. Now:
>>>>>>>1.) Watcher having one of those apps could see right away 
>>>>>>>whether it is possible to communicate
>>>>>>>2.) Composer getting PUBLISHes from 2 sources could 
>>>>>>>immediately know that they are from a similar app
>>>>>>>3.) The app could set its authorization rules using the 
>>>>>>
>>unique id.
>>
>>>>>>>The main downside I can see in this approach that if there 
>>>>>>>are two different proprietary apps that indeed could 
>>>>>>>communicate at least partially, the watcher might make a 
>>>>>>>conclusion that communication is not possible baed on 
>>>>>>>different service-id.
>>>>>>
>>>>This is my concern as well.
>>>>
>>>>
>>>>>>>However, in this case the watcher still
>>>>>>
>>>>>>>would have the characteristics visible and could determine 
>>>>>>>that some interworking could be achieved. 
>>>>>>
>>>>>>Personally, I don't think it is a good idea that an end-user 
>>>>>>should make
>>>>>>such decission. If I get a presence document that indicates 
>>>>>>that another
>>>>>>person is able to receive some type of communication, I 
>>>>>>expect that that
>>>>>>type of communication would actually work when I use it.
>>>>>
>>>>With high probability. There are no guarantees with presence. For 
>>>>instance see my mention above of codecs.
>>>>
>>>>
>>>>
>>>>>Exactly right. If you are an application who does not 
>>>>
>>>>necessarily aim for interoperability, but still uses sip:, 
>>>>INVITE etc., you should be able to explicitely say so.
>>>>
>>>>And you can, without a special notion of service.
>>>>
>>>>
>>>>
>>>>>>>For this kind of 
>>>>>>>reasons there should be careful text about when to use 
>>>>>>>service-id in the first place, and what can be 
>>>>>>
>>determined from it.
>>
>>>>>>>Does someone think that this is an absolutely bad idea? If 
>>>>>>>so, how would you envision solving the issues discussed above?
>>>>>>
>>>>I still don't buy this.
>>>>
>>>>
>>>>
>>>>>>I think more care should be taken in mapping SIP UA 
>>>>>>capabilities into actual
>>>>>>services. That is not so clear to me!
>>>>>
>>>>>The idea is nice, but I don't see a problem in _addition_ 
>>>>
>>>>defining the application with something like 
>>>>"com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind 
>>>>of profile would have more meaning than the list of characteristics.
>>>>
>>>>I am still opposed to defining an attribute with no semantics.
>>>>We have been around this so many times that I am quite 
>>>>certain that it 
>>>>will never be possible to get a useful agreement on what 
>>>>"service" means.
>>>>
>>>>If it is introduced, I expect it to have problems like those 
>>>>of the INFO 
>>>>method. (The arguments for introducing it seem similar.)
>>>>
>>>>	Paul
>>>>
>>>
>>>
>>>_______________________________________________
>>>Simple mailing list
>>>Simple@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/simple
>>>
>>
> 
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	Presence Data Model: Identifying services
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Jonathan Rosenberg wrote:
> 
>> Why? There is no particular reason why tel: should mean voice. We 
>> don't restrict sip: that way, and http: doesn't say much about what 
>> you can do either. (I can use http to exchange html, vxml, xcap, etc.)
>>
>> AFAIK, tel: is an addressing format and namespace. The following is 
>> from  draft-ietf-iptel-rfc2806bis-09:
>>
>>    The "tel" URI telephone number is not restricted in the type of
>>    termination point it refers to.  The termination point can be in the
>>    public telephone network, a private telephone network or the
>>    Internet. The termination point can be fixed or wireless and address
>>    a fixed wired, mobile or nomadic terminal.  The terminal addressed
>>    can support any electronic communication service (ECS) including
>>    voice, data and fax.
> 
> 
> Sigh... here is another case where the tel URI's confusion about what it 
> really is, is getting us into trouble.
> 
> If you think about the tel URI as a urn scheme, whereby resolution to an 
> actual set of communication resources is done using a resolution service 
> (ala DDDS), then it becomes fairly obvious (to me at least) that you 
> would simply never even put that URN into the <contact> of a presence 
> document, since it doesnt represent a communications service at all. If 
> anything, it's closer to a presentity identifier. Rather, you would put 
> the individual service URIs that result from the DDDS lookups.
> 
> However, the tel URI is somewhat schizophrenic. Its not a URN, even 
> though it can use (but doesnt require) URN-like resolution services such 
> as enum. It's like a URN in that it can refer to an abstract name and be 
> resolved to a wealth of different URIs of many different schemes and 
> services. Its unlike a URN in that it is also used to refer to telephone 
> resources, and the implication of the tel URI is that "I can dial this 
> from a PSTN phone or enterprise phone and reach something". In that 
> aspect, its more like a protocol URI.
> 
> What does this mean for presence? To be honest, in the discussion in the 
> data model draft, I have assumed the protocol URI meaning. That is, it 
> refers to a PSTN (or private network) circuit endpoint, period. That, of 
> course, is a BIG assumption, and something we should discuss. I would 
> propose that we keep this assumption and merely state it. In such a 
> case, it would not make sense for the recipient of a presence document 
> to look up  such a tel URI in enum. Rather, it knows its a telephony 
> resource right away and can "dial it".

This feels wrong to me.

I think it is better viewed as "this CAN be reached via the PSTN, and 
MAY also be reachable other ways". At the very least, if I choose to use 
this published address, and there are ENUM entries for it, I should be 
free to use them (potentially making an e2e sip call).

 From a practical perspective, I think this means that if all I have is 
a PSTN phone, then I can consider this contact as one to try. If I have 
a sip device that supports tel uris and ENUM and has a pstn gateway, 
then I can also consider trying this contact. The tuple with this 
contact could show support for all sorts of media if it wants, and if my 
device supports those media then it might try them as well.

	Paul


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Aki Niemi wrote:
> Inline.
> 
> On Fri, 2004-09-24 at 16:59, ext Paul Kyzivat wrote: 
> 
>>>Hmm... If the scheme is that ambiguous, then perhaps its definition
>>>needs to improve.
>>
>>Why? There is no particular reason why tel: should mean voice. We don't 
>>restrict sip: that way, and http: doesn't say much about what you can do 
>>either. (I can use http to exchange html, vxml, xcap, etc.)
> 
> 
> There is a big difference. I can safely assume that for sip:, I am able
> to send a SIP request and the other end will understand it; but probably
> not an HTTP request, or let's say an SMTP message using that URI.
> 
> I admit tel is more problematic, since it is more of a name, really
> without a protocol attached to it, and sending a SIP request to it might
> actually work (because we've defined what that means). But the inverse
> isn't true, since POTS phones don't do IM or anything fancy like that.
> That's why I think it's futile to build support for them on top of tel.
> 
> BTW, there is also really no way to know whether I can send an SMS to
> that telephone number (outside of recognizing it as a mobile number).
> Which is one of the reasons an explicit smsto URI would be nicer...

Well, you could advertise support for it in your presence document. :-)

>>AFAIK, tel: is an addressing format and namespace. The following is from 
>>  draft-ietf-iptel-rfc2806bis-09:
>>
>>    The "tel" URI telephone number is not restricted in the type of
>>    termination point it refers to.  The termination point can be in the
>>    public telephone network, a private telephone network or the
>>    Internet. The termination point can be fixed or wireless and address
>>    a fixed wired, mobile or nomadic terminal.  The terminal addressed
>>    can support any electronic communication service (ECS) including
>>    voice, data and fax.
> 
> Right, but I definitely don't think that IM or voice+IM session are
> ECSs. SIP happens to have a linkage in that if I INVITE a tel URI with
> appropriate media (e.g., voice or fax), then it may work if there is an
> appropriate GW in place or a termination point that speaks SIP.

Why not a SIMPLE-SMS gateway?

>>>IMO the most obvious solution here is that if the endpoint wants to
>>>advertize services other than (or in addition to) telephony, it
>>>publishes a SIP or IM tuple as well as a tel tuple. Is there a
>>>reason this would not work?
>>
>>I don't see a problem here, and so I don't feel a need for a solution.
>>
>>I agree that other address forms are probably desirable in this case, 
>>for a variety of reasons. But that doesn't mean we should
>>automatically assume tel: implies voice telephony and only that.
> 
> I think we have to assume tel implies some telephony, or ECS service
> (whatever that means). If a presentity wants to advertize services
> beyond those, then it is only reasonable that those other contacts are
> also shown in presence.

I think it is reasonable to assume that a tel: address is reachable via 
the PSTN, and hence presumably supports some media usable in that way. 
But it may also support other media when accessed via a protocol (like 
SIP) that supports them.

>>One place where this situation may well come up is when the tel: uri
>>is 
>>used as the presentity address. (Something that should be ok, and
>>makes 
>>sense to use if you are a client and that is all you have.) It may be 
>>that the user doesn't want to expose device addresses, and so provides
>>only the presentity address back as the contact in the tuple(s).
> 
> 
> I don't understand what the problem there is. Even if tel URI was
> allowed as the entity attribute in PIDF (which it isn't AFAIK)

I'm not sure.

The PIDF xml schema says that it can be anyURL. The description of the 
<presence> element says: "The value of the 'entity' attribute is the 
'pres' URL of the PRESENTITY publishing this presence document." But 
AFAIK "is" isn't normative, so the schema takes precedence. I know use 
of sip: urls is common, so I see no reason why tel: couldn't also be used.

Of course there would be the issue of discovering the presence server 
when using a tel: uri as a presentity address. If I know how to use sip 
for presence, then an enum mapping from the tel address to a sip address 
would do the trick. But then I suppose the PS would use the sip address 
as the entity.

 > then that
> presence document would simply state the status of telephony services,
> nothing more.

Why? It can describe everything that is available. And it can use the 
tel address in the tuples if it likes.

	Paul


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Subject: Re: meaning of tel URI in presence data model, was: Re: [Simple]
	Presence Data Model: Identifying services
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> 
> This feels wrong to me.

Agreed.

> 
> I think it is better viewed as "this CAN be reached via the PSTN, and 
> MAY also be reachable other ways". At the very least, if I choose to use 
> this published address, and there are ENUM entries for it, I should be 
> free to use them (potentially making an e2e sip call).

To rephrase slightly: the tel: URI simply says "this is a service that 
is identified by an E.164 number" and adds a level of indirection to 
allow a separate instance to do the binding to one or more instances 
implementing this service, where the binding resolution is performed 
either at the outbound proxy (if used in a SIP device, using any method 
including but not limited to ENUM) or the end system (using ENUM or 
direct dial).

It probably helps if that number is indeed dialable (with number 
translation into a dial string) by a regular, analog phone controlled by 
  some TAPI interface. The usual phone number caveats apply: it may 
whistle at you, it may be an SMS-only number, etc.

I don't see why this level of indirection is problem or requires further 
special-casing within the presence data model.


> 
>  From a practical perspective, I think this means that if all I have is 
> a PSTN phone, then I can consider this contact as one to try. If I have 
> a sip device that supports tel uris and ENUM and has a pstn gateway, 
> then I can also consider trying this contact. The tuple with this 
> contact could show support for all sorts of media if it wants, and if my 
> device supports those media then it might try them as well.
> 
>     Paul
> 
> 
> _______________________________________________
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Markus,

I largely agree with the comments that Jonathan made.

He suggested that the problem at hand is that your PTT isn't really sip 
if it can't interoperate with a regular sip device. That may be a bit 
extreme, but not too far off.

As I suggested in an earlier reply, you could model this by indicating 
that you don't support voice, and instead describe some other capability.

Better yet would be to actually support voice. How hard would it be to 
create a voice:ptt transcoder. Then the situation would be much like a 
codec mismatch. I don't really imagine you would do this right now, but 
that may be the long term path to true interoperability.

More comments below.

	Paul

Markus.Isomaki@nokia.com wrote:
> Hi Paul,
> 
> Thanks for your comments. 
> 
> I guess what we both want to accomplish is this:
> 
> Allow the watcher to better understand whether and how it is able to communicate with a "service" or "communication mean" that is advertised by the presentity. 
> 
> The current approach is to list as many characteristics and parameters of the service as possible, so that the watcher has as much information as possible. In theory this works well, as long as there is some agreed set of characteristics for each possible service. In practice there are no guarantees, since some essential info, such as codec support, is probably discovered only by trial and error. The concept still works well as long as the probability of a "false positive" is reasonably low.
> 
> One idea might be that the presentity should be able to declare some capabilities as mandatory/essential, i.e. saying like "if you don't support at least capabilities A and B, please don't contact me, since there is no value in this from my point of view". An example might be PTT, where the presentity could declare support for "voice", "half-duplex", "INVITE, ..." and floor control "foo", and might insist that even if the watcher supported the same except floor control "bar", the communication should not be attempted. Obviously it would be in the power of the clients and users to decide what to declare as essential and whether to obey those declarations, but at least it would give a hint whether things would work in a reasonable way or not. The ability to declare something as mandatory/essential should apply also to characteristics not understood by the watcher, e.g. if the watcher in the example is a normal VoIP client able to do half-duplex but with no understanding of a
ny type of floor control (except maybe on layer 9 ;-), it could still see that "some feature I don't understand seems to be mandatory, so there is a high probability that the communication will not work". In this way the evil proprietary client could hint "please don't call me even though I'm addressable with SIP unless you know what xyz and zyx are, since if you don't, the outcome may not be anything useful".

I fear where this approach might lead.

I think we will then end up with complex predicates.
(We have more than enough trouble that way with callerprefs, and we 
don't have a comparable description of calleeprefs.)

This would be especially problematic if it meant that the presence UI 
needed to display these constraints. (It would be slightly less bad if 
the subscriber application just applied the constraints to the 
capabilities it knows are available locally in order to filter what is 
displayed.)

> Is there any chance of adding this kind of mechanism? I guess it would be some kind of attribute that would apply to all elements describing service characteristics. I believe it should be part of the data model, so that the designers of the proprietary extensions would know how to use it. 

I really hope we don't need to go there.

> The hard part I see is keeping the standardized characteristics up-to-date and useful. For instance adding "floor control protocol" etc.

> Markus
> 
> 
> 
>>-----Original Message-----
>>From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>Sent: 23 September, 2004 18:47
>>To: Isomaki Markus (Nokia-TP/Espoo)
>>Cc: Stefan.Goeman@siemens.com; simple@ietf.org
>>Subject: Re: [Simple] Presence Data Model: Identifying services
>>
>>
>>I have comments both to Markus' original posting and to the ensuing 
>>discussion. So I will just insert inline in the latest 
>>response I have.
>>
>>	Paul
>>
>>Markus.Isomaki@nokia.com wrote:
>>
>>>Hi Stefan,
>>>
>>>Comments back to you inline. 
>>>
>>>(Yes, I absolutely agree with you on the usefulness of the 
>>
>>Presence Data Model draft and that it should have been the 
>>starting point, at least right after PIDF.)
>>
>>>
>>>>-----Original Message-----
>>>>From: simple-bounces@ietf.org 
>>>>[mailto:simple-bounces@ietf.org]On Behalf
>>>>Of ext Goeman Stefan
>>>>Sent: 23 September, 2004 14:42
>>>>To: 'simple@ietf.org'
>>>>Subject: RE: [Simple] Presence Data Model: Identifying services
>>>>
>>>>
>>>>Hello,
>>>>
>>>>Comments inline
>>>>
>>>>P.S.: In general, I find the Presence Data model draft a good 
>>>>document. This
>>>>is how things should have started in the first place.
>>>>
>>>>Greetings,
>>>>Stefan.
>>>>
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: simple-bounces@ietf.org 
>>>>>[mailto:simple-bounces@ietf.org] On Behalf Of 
>>>>
>>>>Markus.Isomaki@nokia.com
>>>>
>>>>
>>>>>Sent: donderdag 23 september 2004 10:10
>>>>>To: simple@ietf.org
>>>>>Subject: [Simple] Presence Data Model: Identifying services
>>>>>
>>>>>
>>>>>Hi all,
>>>>>
>>>>>I have a comment on Presence Data Model draft about the 
>>>>>identification/characterization of services. (I know this has 
>>>>>been discussed for a while already, but I still think the 
>>>>>current way is somewhat unadequate.)
>>>>>
>>>>>The current Data Model draft says:
>>>>>
>>>>>  In this model, services are not explicitly enumerated.  That is,
>>>>>  there is no "service" attribute with values of "ptt" or 
>>>>>"telephony".
>>>>>  Rather, the service is identified in one of two ways.  In 
>>>>>many cases,
>>>>>  the URI scheme is a clear indicator of service.  An "sms" 
>>>>>URI clearly
>>>>>  indicates SMS, and a "tel" URI clearly indicates 
>>>>>telephony. 
>>>>
>>I don't find the scheme as clearcut as that. I don't think one should 
>>assume "tel" indicates telephony. It perhaps does imply 
>>reachability via 
>>various telephony protocols for purpose of voice 
>>communications, but is 
>>not exclusive to that. If the tel: URI can be mapped to a sip 
>>endpoint 
>>and protocol, then non-voice, non-telephony capabilities are 
>>not excluded.
>>
>>For instance, it may prove convenient to publish a tel URI for a sip 
>>endpoint that is capable of both voice and IM. Then calls via 
>>the PSTN 
>>will work, calls from a sip audio phone will work, calls from another 
>>voice+im client will have access to voice and IM.
>>
>> >>> For some
>>
>>>>>  URIs, there may be many services available, for example, 
>>>>
>>>>SIP.  For
>>>>
>>>>
>>>>>  those services, each service has a set of 
>>>>
>>>>characteristics, each of
>>>>
>>>>
>>>>>  which has a well-defined meaning, such that a system can
>>>>>  unequivocally determine whether or not the service has that
>>>>>  characteristic.  This is discussed in more detail in [5]. 
>>>>>
>>>>>I agree that the contact URI scheme is the most important 
>>>>>piece of information to distinguish what is the "service". 
>>>>>(There are some gaps here too, since e.g. the URI schemes for 
>>>>>SMS or MMS do not even exist, and there may be other non-IETF 
>>>>>protocols that face the same problem.) However, I'm not 
>>>>>convinced that listing the signaling/media characteristics of 
>>>>>the end-point or service really gives enough information to 
>>>>>the watcher to really determine if it can communicate with 
>>>>>the advertised service/application. 
>>>>>
>>>>>The refenced draft [5] has a following example:
>>>>>
>>>>>5.6 Walkie-talkie
>>>>>
>>>>>  The walkie-talkie service allows real-time voice communication
>>>>>  between participants.  Only one participant can speak at a 
>>>>>time; that
>>>>>  is, communication is half-duplex.  Typically, 
>>>>
>>>>participants press a
>>>>
>>>>
>>>>>  button to indicate that they are ready to speak, although other
>>>>>  mechanism (e.g.  voice activation) are occasionally used.
>>>>>
>>>>>  Support for the Walkie-Talkie service can be deduced by 
>>>>>observing the
>>>>>  presence of a contact URI with a scheme of "sip:", 
>>>>
>>>>associated with
>>>>
>>>>
>>>>>  the following minimal set of capabilities: <audio>true</audio>
>>>>>  <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>>>>
>>>>>Presumably this is the same as the service sometimes called 
>>>>>Push-to-Talk (PTT, PoC). The concrete problem is that there 
>>>>>are several _non-interworking_ variants of this service, that 
>>>>>still all would match the characteristics listed above, as 
>>>>>the _SIP_ part might still be standard, at least for some 
>>>>>methods etc. But still the communication would fail even if 
>>>>>the tuple looks OK.
>>>>
>>>>I don't think I understand you here. I would expect that if 
>>>>both end-point
>>>>SIP US support exactly the same methods, the SIP UA should 
>>>>also be able to
>>>>handle those SIP method. Or, to phase it in another way, both 
>>>>end-point
>>>>should be able to communicate with each other only based on 
>>>>using these
>>>>capabilities.
>>>>
>>>
>>>
>>>To some extent that might be possible, but there might be 
>>
>>differences e.g. in floor control etc., which make the 
>>service still practically unusable, which means that the 
>>presence document shouldn't give the impression that the 
>>communication will work. Obviously it is possible to list all 
>>those additional characteristics as well (and this might be a 
>>good idea too), but for instance if presence of a network 
>>game is published, it is easier to _explicitely_ tell what 
>>the game is by providing some sort of unique id which would 
>>be only meaningful to those similar game clients.
>> >
>>
>>>>>(Also if I saw the characteristics above, 
>>>>>I could as well determine that they describe normal VoIP 
>>>>>application running in (very) some old PC with half-duplex 
>>>>>audio drivers, so I claim the service description is hard to 
>>>>>make unambiguous in the first place.)  
>>>>
>>I think the goal should be that all of these are at least minimally 
>>interoperable. I ought to be able to call a PTT contact from 
>>a regular 
>>voice phone and have *something* reasonable happen. (Maybe it is just 
>>oneway voice, with the non-ptt caller never getting the 
>>floor, but that 
>>may be better than nothing.
>>
>>I think the real problem in this case is not with the capability 
>>approach. It is that the 'duplex' capability is 
>>underspecified. There is 
>>no explanation of how a pair of half duplex endpoints (or one 
>>half-duplex and one full-duplex endpoint) agree on who can talk.
>>
>>In the case of the 'voice' capability, it is understood that 
>>having both 
>>endpoints support voice isn't enough to support 
>>communication. There is 
>>an SDP negotiation of codecs, etc. that either succeeds or 
>>fails. There 
>>should be something similar for half-duplex.
>>
>>
>>>>I also have the impression that it will be difficult to 
>>>
>>determine the
>>
>>>>services merely based on the SIP methods (and ...) supported 
>>>>by the SIP UA.
>>>>It possibly might mean that the SIP-based ptt-service on my 
>>>>mobile phone
>>>>might be mapped onto some communication service on your PC. 
>>>>Then, I wonder
>>>>who will be responsible for such kind of mapping? Only the 
>>>>end terminals, or
>>>>is support from the network, presence server, necessary? And, 
>>>>how will one
>>>>service in one domain be mapped into a service in another 
>>>>domain? I believe
>>>>there are some open issues here. 
>>>
>>The intent is that capabilities have common meanings in all 
>>domains, so 
>>they don't need to be mapped. I don't see why there should be any 
>>binding between services and domains.
>>
>>In the general case one should assume that there are devices 
>>of widely 
>>varying capabilities in every domain, and that they know best 
>>what they 
>>are capable of. So the device should be the one deciding if a contact 
>>with and advertised set of capabilties is something it wants 
>>to attempt 
>>communication with.
>>
>>In some environment where there is an intermediary in front 
>>of a device 
>>that understands the device and acts on its behalf, then I suppose it 
>>could make these decisions. But I would consider that the odd case.
>>
>>
>>>I think that it is fully OK to describe some rather simple 
>>
>>things using characteristics, but in many cases applications 
>>are more complex.
>>
>>Clearly we have chosen not to describe capabilities if full detail. 
>>(E.g. we don't publish codec support.) Because of this, there is the 
>>possibility that we make a wrong decision about ability to 
>>communicate 
>>and it doesn't get discovered until we try.
>>
>>This is ok when the expected probability of failure is low. (We 
>>generally expect that there will be some codec in common, or that one 
>>end or the other will be able to introduce a transcoder.)
>>
>>
>>>>>I think the main point of a service tuple is to give the 
>>>>>watcher enough information to know whether he can really 
>>>>>communicate & interoperate with the advertised service. Given 
>>>>>that many services taht are using SIP also have propritary or 
>>>>>non-SIP features, I don't think the current approach is enough. 
>>>>
>>In cases where the existing capabilities aren't sufficient to 
>>ensure a 
>>good probability of communication, then we should indeed 
>>consider adding 
>>additional capabilities.
>>
>>But I continue to object to the notion of "service" as the way to do 
>>that, because it conveys no semantics about what the 
>>capability is that 
>>it describes.
>>
>>
>>>>Do you have an example of a service that uses SIP and also has non
>>>>SIP-features? If your service also uses non-SIP features, 
>>>
>>you can not
>>
>>>>determine the type of service by only looking at the SIP 
>>>
>>capabilities
>>
>>>>supported by the SIP UA. Some indication of that non-SIP 
>>>>feature should also
>>>>be present in the presence document. 
>>>>
>>>
>>>
>>>True, and since some of those non-SIP features might not be 
>>
>>based on (IETF) standards, it might be good to have the 
>>possibility for the developers of such applications to have a 
>>short cut of saying what the app really is, rather than 
>>having to list characteristics. In the end, some applications 
>>are really not even meant to work with any other than the 
>>exactly same application, for instance some networking games. 
>>
>>If the intent is to define a closed world, where 
>>interoperability is not 
>>even considered a desirable feature, then I agree. I think that may 
>>indeed be the case for proprietary games.
>>
>>But for something like PTT, while it might be in the best interest of 
>>the maker of a particular PTT solution to keep it closed, I 
>>don't think 
>>it is in the best interest of society or the IETF that this 
>>happen. This 
>>is precisely the forum in which we try to prevent that from happening.
>>
>>
>>>>>Of course each of these proprietary services are allowed to 
>>>>>define additional status or tuple-level extensions to PIDF.
>>>>
>>>>Yes, you can always extend the PIDF. My comment here is that 
>>>>this might lead
>>>>to an "explosion" of applications or services. It will 
>>>>certainly not help
>>>>interoperability.
>>>
>>My point precisely.
>>
>>
>>>And who would standardize such characteristic definitions. 
>>
>>It has already taken a couple of years to standardize even 
>>"prescaps", which is just a starting point. 
>>
>>>
>>>>>However, I would like to have some kind of "service 
>>>>>identifier" as part of the basic framework so that this could 
>>>>>be done in a consistent manner. I think it would help 
>>>>>especially in making of authorization and composition rules 
>>>>>more simple. The current "class" attribute is way too 
>>>>
>>>>loosely defined.
>>>>
>>>>
>>>>>So the concrete proposal is to include in RPID a "service id" 
>>>>>element that would have a vendor-specific namespace, similar 
>>>>>to e.g. vendor-specific XCAP AUIDs. So for instance the 
>>>>>SIP-based PTT app from vendor xyz.com would have, e.g. 
>>>>><service-id>com.xyz.ptt</service-id>
>>>>>while the PTT app compliant with OMA would have, e.g.
>>>>><service-id>com.openmobilealliance.ptt</service-id>
>>>>
>>I am not fond of this solution. Done this way, we have defined an 
>>element that has no semantics at all, only syntax.
>>
>>I would rather see vendor specific extensions to PIDF. At 
>>least then the 
>>people that support an extension will understand what it means.
>>
>>But I would rather see people come forward and do the hard work of 
>>publicly defining new capabilities so that independent 
>>implementations 
>>can use them in an interoperable way.
>>
>>
>>>>In general, I could agree with that approach. However, I 
>>>>don't see a reason
>>>>why the vendor specific SIP-based PTT should be different 
>>>
>>>>from the OMA based
>>>
>>>>PTT. Simply define one PTT, based on the SIP UA capabilities.
>>>>
>>>
>>>
>>>Well, for instance Nokia has a prorietary PTT application, 
>>
>>which uses SIP INVITE etc., so users are essentially 
>>addressed with sip: URI. However, it has some floor control, 
>>media etc. features which make it basically non-interoperable 
>>with clients that are not specifically following that 
>>proprietary spec. 
>>
>>This is a problem, not a good thing. I realize it has to 
>>happen as part 
>>of the development process, but the goal should be to get out 
>>of it as 
>>fast as possible.
>>
>>I would hope that endpoints supporting this would be 
>>described as well 
>>as possible using existing capabilities, and that a 
>>reasonable effort be 
>>made to do something reasonable when some other sip device connects.
>>
>>If you really can't deal with a regular voice caller at all, then I 
>>would suggest, as a transitional step, saying that you don't support 
>>voice, and then advertising some proprietary capability as an 
>>alternative to voice.
>>
>>
>>>OMA PTT is closer to standard IETF SIP. Unless there is 
>>
>>some conversion, it does not work with Nokia PTT directly. 
>>There should be some way for distinguishing these two in a 
>>presence document, so that there won't be confusion. I think 
>>the current idea is to do some kind of proprietary PIDF 
>>extensions, but in my opinion a standard XML element in RPID 
>>to tell this kind of information would be better, also for 
>>the purposes of authorization rule making.
>>
>>See above for what I think you could do for now. The same issues may 
>>apply to OMA PTT, and a similar solution could be applied to it.
>>
>>Hopefully somebody will come up with a PTT that can be dealt 
>>with more 
>>gracefully. I think that work should be done on better defining 
>>half-duplex, so that it can be used in conjunction with voice to 
>>describe PTT. But it may just turn out to be the wrong thing. 
>>Maybe we 
>>need a "floor-control" capability.
>>
>>
>>>>>in _addition_ to describing sip:, half-duplex audio, and the 
>>>>>supported methods. Now:
>>>>>1.) Watcher having one of those apps could see right away 
>>>>>whether it is possible to communicate
>>>>>2.) Composer getting PUBLISHes from 2 sources could 
>>>>>immediately know that they are from a similar app
>>>>>3.) The app could set its authorization rules using the unique id.
>>>>>
>>>>>The main downside I can see in this approach that if there 
>>>>>are two different proprietary apps that indeed could 
>>>>>communicate at least partially, the watcher might make a 
>>>>>conclusion that communication is not possible baed on 
>>>>>different service-id.
>>>>
>>This is my concern as well.
>>
>> >>> However, in this case the watcher still
>>
>>>>>would have the characteristics visible and could determine 
>>>>>that some interworking could be achieved. 
>>>>
>>>>Personally, I don't think it is a good idea that an end-user 
>>>>should make
>>>>such decission. If I get a presence document that indicates 
>>>>that another
>>>>person is able to receive some type of communication, I 
>>>>expect that that
>>>>type of communication would actually work when I use it.
>>>
>>With high probability. There are no guarantees with presence. For 
>>instance see my mention above of codecs.
>>
>>
>>>Exactly right. If you are an application who does not 
>>
>>necessarily aim for interoperability, but still uses sip:, 
>>INVITE etc., you should be able to explicitely say so.
>>
>>And you can, without a special notion of service.
>>
>>
>>>>>For this kind of 
>>>>>reasons there should be careful text about when to use 
>>>>>service-id in the first place, and what can be determined from it.
>>>>>
>>>>>Does someone think that this is an absolutely bad idea? If 
>>>>>so, how would you envision solving the issues discussed above?
>>>>
>>I still don't buy this.
>>
>>
>>>>I think more care should be taken in mapping SIP UA 
>>>>capabilities into actual
>>>>services. That is not so clear to me!
>>>
>>>The idea is nice, but I don't see a problem in _addition_ 
>>
>>defining the application with something like 
>>"com.openmobilealliance.ptt" or "com.nokia.ptt", if that kind 
>>of profile would have more meaning than the list of characteristics.
>>
>>I am still opposed to defining an attribute with no semantics.
>>We have been around this so many times that I am quite 
>>certain that it 
>>will never be possible to get a useful agreement on what 
>>"service" means.
>>
>>If it is introduced, I expect it to have problems like those 
>>of the INFO 
>>method. (The arguments for introducing it seem similar.)
>>
>>	Paul
>>
>>
> 
> 
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From simple-bounces@ietf.org  Fri Sep 24 17:13:49 2004
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Nancy Greene (QC/EMC) wrote:
> Paul said:
> 
>>I think the real problem in this case is not with the capability 
>>approach. It is that the 'duplex' capability is underspecified. There is 
>>no explanation of how a pair of half duplex endpoints (or one 
>>half-duplex and one full-duplex endpoint) agree on who can talk.
> 
> 
>>the case of the 'voice' capability, it is understood that having both 
>>endpoints support voice isn't enough to support communication. There is 
>>an SDP negotiation of codecs, etc. that either succeeds or fails. There 
>>should be something similar for half-duplex.
> 
> 
> I have been thinking about this too. When it is just a pair of terminals, each could use their own mechanism to not start an RTP session if it detects that the other end has started one first.

If I understand you, that isn't half duplex - it is simplex with the 
direction negotiated. I suppose you could use reinvites to request and 
transfer the floor, but I don't imagine that would work very well.

> But don't we need the duplex info as an attribute in the sdp as well as in a feature tag? It may only apply to one of the media types negotiated.

Probably. Its partially there (in the form of sendonly/recvonly) if you 
want to manage the floor using offer/answer (ugh!). Otherwise something 
else is needed in order to negotiate the need for floor control.

I think the conferencing work also has mechanisms to negotiate the 
establishment of a floor control channel, but I don't think it is SDP.

As long as there is some way of negotiating the floor control in an 
offer/answer, a PTT endpoint can refuse a call that fails to do so. Then 
the incompatibility is comparable to codec incompatibility.

	Paul


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Subject: Re: [Simple] Presence Data Model: Identifying services
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I think this is an overstatement.

To keep it simple, a sip client with only a speaker and microphone can not 
effectively participate in a service which involves a video stream, even to 
be just the receiver.  He can attempt a media negotiation, but it will fail.

Similarly, if the PTT floor control is handled in the RTP / STCP (as some 
of the proposals do), then the SIP is compliant, but the negotiation 
(assuming that the SDP properly represents the details) will fail.

To call this "not SIP" seems very odd.  The SIP would be fully compliant.

Now, one can view the presence properties as a hint, and say "sometimes it 
is not enough information".  In fact, for at least some cases we need to 
say that.
But we should not pretend that the contact URI always properly (much less 
unambiguously) identifies the service.

Yours,
Joel M. Halpern

At 04:34 PM 9/24/2004 -0400, Paul Kyzivat wrote:
>Markus,
>
>I largely agree with the comments that Jonathan made.
>
>He suggested that the problem at hand is that your PTT isn't really sip if 
>it can't interoperate with a regular sip device. That may be a bit 
>extreme, but not too far off.
>
>As I suggested in an earlier reply, you could model this by indicating 
>that you don't support voice, and instead describe some other capability.
>
>Better yet would be to actually support voice. How hard would it be to 
>create a voice:ptt transcoder. Then the situation would be much like a 
>codec mismatch. I don't really imagine you would do this right now, but 
>that may be the long term path to true interoperability.
>
>More comments below.
>
>         Paul


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> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Paul Kyzivat
> Sent: 24.September.2004 19:20
> To: 'simple@ietf.org'
> Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
> Chunking
>=20
>=20
> Out approach to chunking has evolved thru the versions, and=20
> the document
> doesn't seem wholely clear and consistent about what we have=20
> arrived at.
> In fact, I am not entirely certain I know fully what we have decided,
> and I can't figure it out for certain from the document. I=20
> would propose
> specific wording changes, but first I want to get clarification about
> how we think it should now work.
>=20
> Based on scattered information in the doc, and reading between the
> lines, and factoring in stuff from conversations along the way, I
> *think* the following is intended:
>=20
> - only SEND messages are chunkable
>=20
> - there are two kinds of chunks: interrruptible and uninterrupible.
>=20
> - a chunk is uninterruptible if it has a Byte-Range header with a
>     numeric range-end.
>=20
> - other chunks are interruptible. (No Byte-Range, or a Byte-Range
>     with a range-end of "*".)
>=20
> - uninterruptible chunks MUST NOT have a range-end greater than 2048.
>=20
> - chunks other than the last SHOULD (or MUST?) have a body ('data'
>     in the syntax) of length exactly 2048. If SHOULD, what is a
>     valid reason to violate?
>=20
> - responses never have bodies at all. Requests other than SEND
>     may have a body but are unchunkable. The presence of a Byte-Range
>     in a request other than SEND says nothing about the body of that
>     request. (In REPORT it refers to the SEND being reported on.
>     See separate thread on Byte-Range.) This at best seems irregular
>     and weird.
>=20
> Section 6.3.1 says: "it is possible the body is shorter than the
> range-end field indicates. This can occur if the sender interrupted a
> SEND request unexpectedly." Based on what I have interpretted above,
> this is no longer possible. (To be interrupted it must have had a
> range-end of "*".)

I think it is still possible, but it means that it was an abnormal =
behaviour. Perhaps we should include some text discussing a recovery =
mechanism for this case.

/Hisham

>=20
>=20
>=20
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> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Paul Kyzivat
> Sent: 24.September.2004 19:21
> To: 'simple@ietf.org'
> Subject: [Simple] Review of draft-ietf-simple-message-sessions-08
> -Overlapping Chunks
>=20
>=20
> Section 6.3.1 discusses how to handle the receipt of=20
> overlapping chunks,
> specifying that the last received copy of any particular byte wins. I
> don't think this is good.
>=20
> If the overlapping chunks contain consistent data then it=20
> doesn't matter
> which takes precedence. If they contain conflicting data then=20
> reordering
> of chunks can affect the result unpredictably. This rule also prevents
> the recipient from rendering a byte when it and all preceding=20
> bytes have
> been received.
>=20
> I think it should be permissible for the recipient to check for
> inconsistency in the overlapping case, and report an error if=20
> so. We may
> want to require this checking. (Since it typically won't happen it
> shouldn't have a performance impact.)

Since it typically won't happen, I would argue that we don't need to =
complicate the protocol to handle it.

/Hisham

>=20
>=20
>=20
> _______________________________________________
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Subject: RE: [Simple] Review of draft-ietf-simple-message-sessions-08 - Byte
	Ranges in REPORTs
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I agree. The situation is further complicated when a relay decides to =
re-chunk. The relay then has to consume the REPORTs and generate it own =
to pass upstream.

/Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Paul Kyzivat
> Sent: 24.September.2004 19:21
> To: simple@ietf.org
> Subject: [Simple] Review of=20
> draft-ietf-simple-message-sessions-08 - Byte
> Ranges in REPORTs
>=20
>=20
> I am really confused about how byte ranges are supposed to work in
> reports, and I suspect we are permitting a bunch of nonsense.
>=20
> Fundamentally, a message can only be considered a success if all its
> bytes are received successfully. If it is to be positively=20
> acknowledged
>    with success reports, then there must be success reports covering
> every byte. Is there any point in sending a success report=20
> for less than
> the entire message? It doesn't help the receiver, and actually makes
> more work because it then requires keeping track of what=20
> bytes have been
> confirmed. And it doesn't ease the job of the sender, who could easily
> just wait until all has been received before sending the=20
> success report.
>=20
> The flip side is that a message fails if a single byte fails. At that
> point it does no good to know that other parts succeeded or failed. So
> while it makes sense to send a failure report based on problems with a
> single chunk, there is little value in saying which byte range failed.
>=20
> Bottom line: unless I have missed something, there is no reason for
> Byte-Range in REPORT or responses. A success report should=20
> only be sent
> after all the chunks of a message have been received. A=20
> failure report,
> or an error status, can be sent at the time the failure of a single
> chunk is known, but need not identify the byte range. (In that case,
> maybe the Byte-Range could be optional for debugging purposes only.)
>=20
>=20
>=20
>=20
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I believe it applies to a message (spread across 1 to n SENDS requests, =
i.e. chunks).

Hisham

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org]On Behalf
> Of ext Paul Kyzivat
> Sent: 24.September.2004 19:21
> To: simple@ietf.org
> Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
> max-size
>=20
>=20
> Section 7.1 says: "An endpoint MAY indicate the maximim size message
> they wish to receive using the max-size a-line attribute".
>=20
> Does this apply only to a *message* - namely the content of a SEND
> request? Or does this apply to any and all requests (including future
> extensions) that have content?
>=20
>=20
>=20
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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From simple-bounces@ietf.org  Mon Sep 27 07:51:51 2004
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From: Helsen Frank <Frank.Helsen@siemens.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
Date: Mon, 27 Sep 2004 13:44:03 +0200
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This is indeed what I had in mind.  Thanks for the clarification.  And I
agree that this is not presence information. The approach in RFC3840 where
the device capabilities are transported in the REGISTER seems the correct
solution.  Better then having it in a NOTIFY.  However, RFC3840 does not
create the option to specify the device.  One would not have to think about
what device information a provider/PS/Application would like to have.  Just
specify which device it is and it is up to him to get the required
information (using device profiles).

Regards,

Frank

Frank Helsen - MPM/IMC System Engineer
Mail: frank.helsen@siemens.com
Tel:+32 14 252573
Mobile: +32 494 692959
Siemens HRT IC MN D D1


-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
Sent: vrijdag 24 september 2004 17:48
To: Paul Kyzivat
Cc: Helsen Frank; 'simple@ietf.org'
Subject: Re: [Simple] Extend UA Capabiity Extension to PIDF


My guess is that Helsen is looking at something equivalent to the UAProf 
functions in WAP, which allow a client to push fairly complicated 
information about itself, such as screen size and JVM. I don't think 
this is "presence" in the sense that much of this information is 
probably not useful to watchers. However, it does represent valid device 
information and could conceivably appear there.

We do have a specification, RFC 3840, which allows a client to declare 
its capabilities to the server. That specification does not dictate how 
this information is used. One usage is caller preferences, RFC 3841, 
which allows for capability-based called routing. However, I think 
Helsen has in mind a usage where the provider/domain owner simply wants 
to look at this profile data for some reason, and that is certainly 
possible.

We have only a very limited set of capabilities declare right now, a 
much, much, much smaller subset of what is in UAProf.

-Jonathan R.



Paul Kyzivat wrote:

> Helsen,
> 
> I don't understand your terminology very well, or what you are trying to 
> achieve.
> 
> By 'client' I guess you mean a device addressed by a contact in a tuple 
> of the presence document, rather than a subscriber to presence. Is that 
> right?
> 
> I don't know precisely what role you have in mind for 'operator'. Is it 
> the operator of the presence server itself? Or is it something that is a 
> client of the PS. What is it doing that requires it to know this 
> information?
> 
> In general the device will probably be publishing its own presence 
> status to the presence server. If so the extension you suggest would 
> simply cause it to push this invariant information over and over.
> 
> This kind of info can already be polled using OPTIONS. Why isn't that 
> enough?
> 
>     Paul
> 
> Helsen Frank wrote:
> 
>> Hello,
>>
>>
>> At this moment, draft-ietf-simple-prescaps-ext-01 does not provide a 
>> way to indicate to the server which client the user uses.  I can 
>> imagine that an operator has specific client capabilities he would 
>> like to know (e.g. java virtual machine version, screen size).  When 
>> including this client information, the server could retrieve the 
>> client capabilities from his e.g. his DB.  With this extension, the 
>> operator has control over the client information and is not dependend 
>> on the information the client provides to him.
>> Next example shows a possible extension
>>
>> <ClientInfo>
>>         <Manufacturor> Siemens </Manufacturor>
>>         <Model> CX65 </Model>
>>         <Version>2.103564</Version>
>> </ClientInfo>
>>
>> Regards,
>>
>> Frank
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
> 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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hisham.khartabil@nokia.com wrote:
> 
>>-----Original Message-----
>>From: simple-bounces@ietf.org 
>>[mailto:simple-bounces@ietf.org]On Behalf
>>Of ext Paul Kyzivat
>>Sent: 24.September.2004 19:20
>>To: 'simple@ietf.org'
>>Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
>>Chunking
>>
>>
>>Out approach to chunking has evolved thru the versions, and 
>>the document
>>doesn't seem wholely clear and consistent about what we have 
>>arrived at.
>>In fact, I am not entirely certain I know fully what we have decided,
>>and I can't figure it out for certain from the document. I 
>>would propose
>>specific wording changes, but first I want to get clarification about
>>how we think it should now work.
>>
>>Based on scattered information in the doc, and reading between the
>>lines, and factoring in stuff from conversations along the way, I
>>*think* the following is intended:
>>
>>- only SEND messages are chunkable
>>
>>- there are two kinds of chunks: interrruptible and uninterrupible.
>>
>>- a chunk is uninterruptible if it has a Byte-Range header with a
>>    numeric range-end.
>>
>>- other chunks are interruptible. (No Byte-Range, or a Byte-Range
>>    with a range-end of "*".)
>>
>>- uninterruptible chunks MUST NOT have a range-end greater than 2048.
>>
>>- chunks other than the last SHOULD (or MUST?) have a body ('data'
>>    in the syntax) of length exactly 2048. If SHOULD, what is a
>>    valid reason to violate?
>>
>>- responses never have bodies at all. Requests other than SEND
>>    may have a body but are unchunkable. The presence of a Byte-Range
>>    in a request other than SEND says nothing about the body of that
>>    request. (In REPORT it refers to the SEND being reported on.
>>    See separate thread on Byte-Range.) This at best seems irregular
>>    and weird.
>>
>>Section 6.3.1 says: "it is possible the body is shorter than the
>>range-end field indicates. This can occur if the sender interrupted a
>>SEND request unexpectedly." Based on what I have interpretted above,
>>this is no longer possible. (To be interrupted it must have had a
>>range-end of "*".)
> 
> 
> I think it is still possible, but it means that it was an abnormal behaviour. Perhaps we should include some text discussing a recovery mechanism for this case.

It is abnormal in the sense that it can only arise if there is a 
violation of the specification. Normally we don't specify corrective 
action in those cases.

If there is reason we can recommend a recovery action (it is pretty 
obvious), but should at least note that this is recovery from incorrect 
sender behavior.

	Paul


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hisham.khartabil@nokia.com wrote:
> I believe it applies to a message (spread across 1 to n SENDS requests, i.e. chunks).

I am inclined to agree. That means that recipients should be prepared to 
receive up to 2k bodies with any other request that is permitted to 
contain a body. (In this spec that is only REPORT.)

I think we just need to make this clear. It wasn't entirely clear to me 
from reading the existing text.

	Paul

> Hisham
> 
> 
>>-----Original Message-----
>>From: simple-bounces@ietf.org 
>>[mailto:simple-bounces@ietf.org]On Behalf
>>Of ext Paul Kyzivat
>>Sent: 24.September.2004 19:21
>>To: simple@ietf.org
>>Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
>>max-size
>>
>>
>>Section 7.1 says: "An endpoint MAY indicate the maximim size message
>>they wish to receive using the max-size a-line attribute".
>>
>>Does this apply only to a *message* - namely the content of a SEND
>>request? Or does this apply to any and all requests (including future
>>extensions) that have content?
>>
>>
>>
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple
>>
> 
> 


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Subject: RE: [Simple] Review of draft-ietf-simple-message-sessions-08 -
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> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 27.September.2004 17:04
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: simple@ietf.org
> Subject: Re: [Simple] Review of=20
> draft-ietf-simple-message-sessions-08 -
> Chunking
>=20
>=20
>=20
>=20
> hisham.khartabil@nokia.com wrote:
> >=20
> >>-----Original Message-----
> >>From: simple-bounces@ietf.org=20
> >>[mailto:simple-bounces@ietf.org]On Behalf
> >>Of ext Paul Kyzivat
> >>Sent: 24.September.2004 19:20
> >>To: 'simple@ietf.org'
> >>Subject: [Simple] Review of draft-ietf-simple-message-sessions-08 -
> >>Chunking
> >>
> >>
> >>Out approach to chunking has evolved thru the versions, and=20
> >>the document
> >>doesn't seem wholely clear and consistent about what we have=20
> >>arrived at.
> >>In fact, I am not entirely certain I know fully what we=20
> have decided,
> >>and I can't figure it out for certain from the document. I=20
> >>would propose
> >>specific wording changes, but first I want to get=20
> clarification about
> >>how we think it should now work.
> >>
> >>Based on scattered information in the doc, and reading between the
> >>lines, and factoring in stuff from conversations along the way, I
> >>*think* the following is intended:
> >>
> >>- only SEND messages are chunkable
> >>
> >>- there are two kinds of chunks: interrruptible and uninterrupible.
> >>
> >>- a chunk is uninterruptible if it has a Byte-Range header with a
> >>    numeric range-end.
> >>
> >>- other chunks are interruptible. (No Byte-Range, or a Byte-Range
> >>    with a range-end of "*".)
> >>
> >>- uninterruptible chunks MUST NOT have a range-end greater=20
> than 2048.
> >>
> >>- chunks other than the last SHOULD (or MUST?) have a body ('data'
> >>    in the syntax) of length exactly 2048. If SHOULD, what is a
> >>    valid reason to violate?
> >>
> >>- responses never have bodies at all. Requests other than SEND
> >>    may have a body but are unchunkable. The presence of a=20
> Byte-Range
> >>    in a request other than SEND says nothing about the body of that
> >>    request. (In REPORT it refers to the SEND being reported on.
> >>    See separate thread on Byte-Range.) This at best seems irregular
> >>    and weird.
> >>
> >>Section 6.3.1 says: "it is possible the body is shorter than the
> >>range-end field indicates. This can occur if the sender=20
> interrupted a
> >>SEND request unexpectedly." Based on what I have interpretted above,
> >>this is no longer possible. (To be interrupted it must have had a
> >>range-end of "*".)
> >=20
> >=20
> > I think it is still possible, but it means that it was an=20
> abnormal behaviour. Perhaps we should include some text=20
> discussing a recovery mechanism for this case.
>=20
> It is abnormal in the sense that it can only arise if there is a=20
> violation of the specification. Normally we don't specify corrective=20
> action in those cases.

No, I meant device or user behaviour that causes the sending to be =
immediately aborted. Nothing to do with protocol violation. (eg: abnorla =
termination of the application).

/Hisham

>=20
> If there is reason we can recommend a recovery action (it is pretty=20
> obvious), but should at least note that this is recovery from=20
> incorrect=20
> sender behavior.
>=20
> 	Paul
>=20
>=20

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hisham.khartabil@nokia.com wrote:
> 
>>>>Section 6.3.1 says: "it is possible the body is shorter than the
>>>>range-end field indicates. This can occur if the sender 
>>>
>>interrupted a
>>
>>>>SEND request unexpectedly." Based on what I have interpretted above,
>>>>this is no longer possible. (To be interrupted it must have had a
>>>>range-end of "*".)
>>>
>>>
>>>I think it is still possible, but it means that it was an 
>>
>>abnormal behaviour. Perhaps we should include some text 
>>discussing a recovery mechanism for this case.
>>
>>It is abnormal in the sense that it can only arise if there is a 
>>violation of the specification. Normally we don't specify corrective 
>>action in those cases.
> 
> 
> No, I meant device or user behaviour that causes the sending to be immediately aborted. Nothing to do with protocol violation. (eg: abnorla termination of the application).

As best I can tell, we are saying that you must decide, when beginning 
to send, whether to use an interruptible chunk, or an uninterruptible 
chunk. If you use an uninterruptible chunk you specify the range-end as 
a number and include that many bytes of body. If you choose an 
interruptible chunk, you use "*" for range-end and then you put as much 
body as you like.

An application that chooses to use an uninterruptible chunk, and then 
fails to include the full number of bytes for it has violated the protocol.

Effectively, an application should only use an uninterruptible chunk if 
it already has all the bytes in hand and ready to send.

	Paul


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Date: Mon, 27 Sep 2004 18:04:14 +0200
From: Goix Walter <Walter.Goix@TILAB.COM>
Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
To: Helsen Frank <Frank.Helsen@siemens.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I also agree that UAProf-type of information should not be carried along
with presence information. I would rather prefer some kind of Profile:
header in SIP requests that points to an OMA-like UAProf URL which also
describes SIP caps in a more complete way than RFC3840 does (I don't
think that RFC has been thought to define such info)

Laurent-Walter Goix
Telecom Italia - TILAB
Service and Platform Innovation - Service Control Platforms

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org] On Behalf Of Helsen Frank
> Sent: Monday, September 27, 2004 1:44 PM
> To: 'Jonathan Rosenberg'
> Cc: Paul Kyzivat; 'simple@ietf.org'
> Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
>=20
>=20
> This is indeed what I had in mind.  Thanks for the=20
> clarification.  And I agree that this is not presence=20
> information. The approach in RFC3840 where the device=20
> capabilities are transported in the REGISTER seems the=20
> correct solution.  Better then having it in a NOTIFY. =20
> However, RFC3840 does not create the option to specify the=20
> device.  One would not have to think about what device=20
> information a provider/PS/Application would like to have. =20
> Just specify which device it is and it is up to him to get=20
> the required information (using device profiles).
>=20
> Regards,
>=20
> Frank
>=20
> Frank Helsen - MPM/IMC System Engineer
> Mail: frank.helsen@siemens.com
> Tel:+32 14 252573
> Mobile: +32 494 692959
> Siemens HRT IC MN D D1
>=20
>=20
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]=20
> Sent: vrijdag 24 september 2004 17:48
> To: Paul Kyzivat
> Cc: Helsen Frank; 'simple@ietf.org'
> Subject: Re: [Simple] Extend UA Capabiity Extension to PIDF
>=20
>=20
> My guess is that Helsen is looking at something equivalent to=20
> the UAProf=20
> functions in WAP, which allow a client to push fairly complicated=20
> information about itself, such as screen size and JVM. I don't think=20
> this is "presence" in the sense that much of this information is=20
> probably not useful to watchers. However, it does represent=20
> valid device=20
> information and could conceivably appear there.
>=20
> We do have a specification, RFC 3840, which allows a client=20
> to declare=20
> its capabilities to the server. That specification does not=20
> dictate how=20
> this information is used. One usage is caller preferences, RFC 3841,=20
> which allows for capability-based called routing. However, I think=20
> Helsen has in mind a usage where the provider/domain owner=20
> simply wants=20
> to look at this profile data for some reason, and that is certainly=20
> possible.
>=20
> We have only a very limited set of capabilities declare right now, a=20
> much, much, much smaller subset of what is in UAProf.
>=20
> -Jonathan R.


Gruppo Telecom Italia - Direzione e coordinamento di Telecom Italia =
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Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Goix Walter <Walter.Goix@TILAB.COM>
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On Mon, 2004-09-27 at 19:04, ext Goix Walter wrote:
> I also agree that UAProf-type of information should not be carried along
> with presence information. I would rather prefer some kind of Profile:
> header in SIP requests that points to an OMA-like UAProf URL which also
> describes SIP caps in a more complete way than RFC3840 does (I don't
> think that RFC has been thought to define such info)

How about using the User-Agent header field?

Cheers,
Aki





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Date: Tue, 28 Sep 2004 14:26:23 +0200
From: Goix Walter <Walter.Goix@TILAB.COM>
Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
To: Aki Niemi <aki.niemi@nokia.com>, ext Goix Walter <Walter.Goix@TILAB.COM>
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I'm not sure I get your point, whether you propose to use the User
Agent: header as is (1), "modified" for transporting a UAProf URI (2),
or if you see some external mechanism that knows the matching between
User-Agent: header value and UAProf-type of capabilities related to SIP
(3)...

If (1), I personally see the User-Agent: header as something related to
SIP UA "binary" module name and version (for vendor's use only, or for
opaque upgrade from an operator when the version is considered as
obsolete). I think the info here is also related to device caps (screen
size, color for max img size when sending an image or things like that).
The User-Agent may not provide that info unless (3), but then you may
add complexity in your network if you need to provision this matching
(version/UAProf) on many NEs. The matching itself may also be tricky in
some situations.

IMHO, I think having this info provided directly by the UA itself could
be helpful and simplier for Application Servers or proxies to process a
request based on such info.

walter
=20
> -----Original Message-----
> From: Aki Niemi [mailto:aki.niemi@nokia.com]=20
> Sent: Tuesday, September 28, 2004 1:59 PM
> To: ext Goix Walter
> Cc: Helsen Frank; Jonathan Rosenberg; Paul Kyzivat; SIMPLE WG
> Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
>=20
>=20
> On Mon, 2004-09-27 at 19:04, ext Goix Walter wrote:
> > I also agree that UAProf-type of information should not be carried=20
> > along with presence information. I would rather prefer some kind of=20
> > Profile: header in SIP requests that points to an OMA-like=20
> UAProf URL=20
> > which also describes SIP caps in a more complete way than=20
> RFC3840 does=20
> > (I don't think that RFC has been thought to define such info)
>=20
> How about using the User-Agent header field?
>=20
> Cheers,
> Aki
>=20
>=20
>=20
>=20
>=20


Gruppo Telecom Italia - Direzione e coordinamento di Telecom Italia =
S.p.A.

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Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
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To: ext Goix Walter <Walter.Goix@TILAB.COM>
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On Tue, 2004-09-28 at 15:26, ext Goix Walter wrote:
> I'm not sure I get your point, whether you propose to use the User
> Agent: header as is (1), "modified" for transporting a UAProf URI (2),
> or if you see some external mechanism that knows the matching between
> User-Agent: header value and UAProf-type of capabilities related to SIP
> (3)...
> 
> If (1), I personally see the User-Agent: header as something related to
> SIP UA "binary" module name and version (for vendor's use only, or for
> opaque upgrade from an operator when the version is considered as
> obsolete). I think the info here is also related to device caps (screen
> size, color for max img size when sending an image or things like that).
> The User-Agent may not provide that info unless (3), but then you may
> add complexity in your network if you need to provision this matching
> (version/UAProf) on many NEs. The matching itself may also be tricky in
> some situations.

Which I guess leaves you with option (2). There doesn't seem to be much
in the way of putting information there that goes beyond the name and
version of the binary. In web browsers this information many times
contains OS, and even CPU information.

Having said that though, if the intention originally was to have CC/PP
in SIP, then it would've been best to say so to begin with...

> IMHO, I think having this info provided directly by the UA itself could
> be helpful and simplier for Application Servers or proxies to process a
> request based on such info.

I agree, but whether you need something beyond callee-caps and
User-Agent is then another question. 

Also, I think some information actually does belong in presence. For
example, it might be worthwhile expressing maximum attachment size in a
mailto tuple. Depends what you would like to do, and who the primary
consumer of this information would be (e2e vs. e2m).

Cheers,
Aki

> walter
>  
> > -----Original Message-----
> > From: Aki Niemi [mailto:aki.niemi@nokia.com] 
> > Sent: Tuesday, September 28, 2004 1:59 PM
> > To: ext Goix Walter
> > Cc: Helsen Frank; Jonathan Rosenberg; Paul Kyzivat; SIMPLE WG
> > Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
> > 
> > 
> > On Mon, 2004-09-27 at 19:04, ext Goix Walter wrote:
> > > I also agree that UAProf-type of information should not be carried 
> > > along with presence information. I would rather prefer some kind of 
> > > Profile: header in SIP requests that points to an OMA-like 
> > UAProf URL 
> > > which also describes SIP caps in a more complete way than 
> > RFC3840 does 
> > > (I don't think that RFC has been thought to define such info)
> > 
> > How about using the User-Agent header field?
> > 
> > Cheers,
> > Aki
> > 
> > 
> > 
> > 
> > 
> 
> 
> Gruppo Telecom Italia - Direzione e coordinamento di Telecom Italia S.p.A.
> 
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From: Goix Walter <Walter.Goix@TILAB.COM>
Subject: RE: [Simple] Extend UA Capabiity Extension to PIDF
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>=20
> Which I guess leaves you with option (2). There doesn't seem=20
> to be much in the way of putting information there that goes=20
> beyond the name and version of the binary. In web browsers=20
> this information many times contains OS, and even CPU information.
>=20
> Having said that though, if the intention originally was to=20
> have CC/PP in SIP, then it would've been best to say so to=20
> begin with...


I agree that option (2) is not the right way to go.=20
Going UAProf, which may be useful for many services, implies to put
CC/PP in SIP as you said (maybe an URL into an header in some cases, or
the same URL as a XML tag in presence information), and also to extend
CC/PP with SIP-related info

>=20
> > IMHO, I think having this info provided directly by the UA itself=20
> > could be helpful and simplier for Application Servers or proxies to=20
> > process a request based on such info.
>=20
> I agree, but whether you need something beyond callee-caps=20
> and User-Agent is then another question.=20
>=20
> Also, I think some information actually does belong in=20
> presence. For example, it might be worthwhile expressing=20
> maximum attachment size in a mailto tuple. Depends what you=20
> would like to do, and who the primary consumer of this=20
> information would be (e2e vs. e2m).

I do believe as well there is a need to put these kind of caps into
presence info. CC/PP is not the only option for this but could be a
rich, standard way to provision such info from the UA to the server, for
example through an URL into a PUBLISH header, which could also be reused
in other methods.
Once they get this info, presence servers can decide whether and how to
send this info (or part of it) to watchers, by sending an HTTL URL of
the profile into PIDF doc (this woul require watchers to have an HTTP
stack) or extracting info and putting it into some PIDF extension (like
you said "max attachment size" for example, or supported content-types
based on the SIP method or type of session, etc. that are not supported
by callee-caps). In this case, why not reuse CC/PP schema and if not
enough simply extend it? CC/PP already has a lot of info that may be
useful for watchers.

walter=20

>=20
> Cheers,
> Aki


Gruppo Telecom Italia - Direzione e coordinamento di Telecom Italia =
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From simple-bounces@ietf.org  Tue Sep 28 17:14:06 2004
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Subject: [Simple] updated presence data model draft
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Folks,

I've just submitted an updated presence data model draft. Until it 
appears in the archives, you can pick it up as:

http://www.jdrosen.net/papers/draft-ietf-simple-presence-data-model-00.txt

There are a number of changes since the last version, which I will send 
in a separate note. The most important one to note is that, instead of 
using <tuple> to encode device and person information (person is the new 
term for user data, like "in a meeting" and "sad"), it is encoded using 
a separate <device> and <person> element which are defined in the 
document, including schema. The reason for this is:

1. its backwards compatible with existing pidf implementations, whereas 
the former approach would have caused problems
2. its the natural way to do this if you were designing the schema from 
scratch anyway
3. it avoids the need for the "ugliness" of a separate <tuple-type> 
element to tell you what the tuple actually was

The drawback is that it will require modification of the partial 
publication and partial notification drafts. However, as these are still 
I-D it doesnt seem so troubling, as compared to the more real issues above.

This revision doesn't include the latest discussions on tel URI and 
service identification.

Most of the revisions are based on conclusions reached by the design 
team when we met during IETF. In particular, we believe that there is 
sufficient agreement on the document that we can go ahead and update 
rpid, cipid and related spcifications to be in line with the model, and 
proceed with those documents at this time.

There are some open issues in the data model, which I will send out 
notes on shortly. I don't think any of these gate rpid/cipid from 
proceeding. Henning has given me source for those documents, and I'll be 
modifying and submitting updates to those shortly.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: [Simple] diffs in presence data model from previous version
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* Terminology: using "person" to refer to the human-user status (busy, in a
meeting, etc.), with presentity representing the combination of person,
device and service information

* mention that the presentity URI will often be the SIP AOR, and that
such equivalence is recommended.

* UUID URN for device ID

* mention that device IDs are ideally obtained from operating system

* clarified the role of characteristics and status information for
devices - cannot be used to determine nature of the service. Rather,
aids in the selection of service based on what the device
is. Clarified that status of a device does not affect status of the
service unless it is known otherwise.

* cleaned up document to remove references to "what is a tuple", or
"this group has had trouble" and other discussions not appropriate in
an RFC

* clarified that it is not that important to cleanly delineate status
and characteristic information.

* syntax is now using new <device> and <person> elements, rather than
re-using tuple. It was felt that this was likely to be backwards
compatible with existing PIDF implementations, whereas the previous
approach would not be. It does mean that partial publish and partial
notify specs will need to change, but they are not done yet.

* clarified that a PUA can "represent" a service, device or person,
but is usually a service, and the discussion focuses on that.

* changed definition of service, presentity and device views, based on
the fact that a presence document has to contain at least one service
tuple

* updated references and examples

* removed plan-of-action section

* clarified that privacy filtering can run at the request of the
presentity or other parties

* added dsicussion of deciding to combine a set of services together
because they provided contacts that are gruus under the same aor

* added a lot of text on overriding published information and on
time-based conflict resolution in presence servers

* removed example of helpdesk application as a case where the r-uri of
the publish is not the same as the entity field in the presence
document, since its confusing and not a good use case.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From simple-bounces@ietf.org  Tue Sep 28 17:23:19 2004
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Subject: [Simple] Pres Data Model Open Issue #1: in-person communications
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Currently, there is no easy way to incorporate the notion of "in person" 
or "written/postal" communications as services represented in the data 
model. The reason is that there is no URI scheme for them, and no 
attributes that can be used to define them. The issue is, what to do 
about this?

My proposal is to punt the issue in this document. State that it is 
expected that future specifications will address how to represent these.

To actually solve the problem, I think the right approach is to define a 
URN scheme along the lines of "urn:inperson" to represent an in-person 
communications scheme. For postal communications, I think you want a URN 
that includes the postal address, something like 
urn:postal:sname=Lanidex%20Plaza-snumber=600-state=NJ-city=Parsippany-zip=07054 
or something like that. Thats probably a fairly hard problem to solve, 
and I think even less important for us to tackle.

However, we can defer working on either of those in earnest for a bit, 
until our basic work is done.

Comments?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Subject: [Simple] Pres Data Model Open Issue #2: fate of
	roach-simple-service-features?
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Currently, draft-roach-simple-service-features is an expired I-D. The 
approach it describes, of using service attributes rather than an 
enumeration of services, has been incorporated into the model already. 
However, the draft has useful explanatory text and examples. Should this 
draft proceed separately or should it be folded into the data model draft?

I would propose that the discussion of the problems with service 
enumeration be integrated into the data model, along with the benefits 
of service deduction. However, the examples of describing various 
services should probably proceed as a BCP in a separate document.

Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Simple] Pres Data Model Open Issue #3, UUID URN or something else
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One of the things the design team spent a lot of time discussing was 
what a device ID looked like. Our conclusion, as documented in the I-D, 
is that its a URN. We can't lock down the specific type of URN, but for 
things where MAC address is a useful unique device identifier, the UUID 
URN (as it makes use of a MAC address) is a good choice.

However, upon a more detailed read of the UUID URN specification, I am 
not sure we can use it. The reason is that the UUID URN spec 
(http://www.ietf.org/internet-drafts/draft-mealling-uuid-urn-03.txt) 
describes several ways of computing the URN. There are actually four 
versions. The time-based version uses the host MAC, along with the 
current time, to generate the UUID. The name-based version doesnt use 
time, but relies on a unique namespace, such as DNS or URIs, as a seed. 
MAC address is not a choice.

So, the problem is, the UUID form that utilizes the MAC address also 
includes a time-based component. This serves to provide uniqueness, but 
makes it useless as a meaningful identifier that other presence sources 
- like an ethernet switch or 802.11 AP, would be able to provide 
information about.

So, the options as I see them are:

1. use UUID URN anyway, lose the correlation properties we wanted
2. dont specify the URN form at all
3. don't specify the URN form in this document, but work on this problem 
in downstream specifications

I would propose approach (3). In particular, I think the data model 
should revert to being agnostic on the URN scheme, reference the UUID 
URN as one possibility, document its problems. We should develop a 
separate specification that defines guidelines for choosing the URN for 
various devices in order to get the correlation properties we want. I 
also think we should take a stab at defining a few URN schemes that 
would be useful - MAC and ESN in particular.

Comments?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From simple-bounces@ietf.org  Tue Sep 28 17:36:53 2004
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Simple] Pres Data Model Issue #4, indicating the source of data
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The data model draft spends a fair bit of time talking about conflicting 
data, and how a presence server can resolve the conflicts. One of the 
things it talks about is making use of presence meta-data, and in 
particular, the source of a piece of presence data, as a tool in 
deciding how to resolve conflicts.

The question is, do we want to do any work in defining ways of 
identifying the source of the presence information?

I am strongly inclined to say "no", and simply leave this as a potential 
future issue to address. I would modify the data model draft to talk 
about meta-data, but not say anything about future standardization or 
any specific specs where such meta-data are defined.

Comments?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From simple-bounces@ietf.org  Tue Sep 28 17:43:48 2004
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Date: Tue, 28 Sep 2004 17:29:49 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Simple] Pres Data Model Open Issue #5: what does idle mean?
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This isn't really an issue with the data model so much as rpid, but its 
used as an example in the data model.

The question is, in the context of the data model, what does <idle> 
mean? Does <idle> apply to a person? To a device? To a service? For each 
case, what does it mean? Does it merely mean, "the likelihood is that 
attempts to reach this service will result in a no-answer", or is it 
something more specific, such as "idle indicates that no user input has 
been provided to this device (service) recently".

The first question is to determine its definition, I think. Once you 
know what it means, you can decide whether it applies to devices, 
services or person. I'm somewhat on the fence here. One argument is that 
you want a concise and measurable definition, so that a PUA can 
concretely say, "this service is idle", and have the recipient of the 
document know what it means. That would argue for something based on 
user input. The problem with that definition is that ultimately, the 
lack of user input is useful because it is an indicator of the real 
quantity of interest - the likelihood that I'll get a "no answer". So, 
why not define it as the data we really want. The problem is that you 
will see vast differences across devices in declaring themselves idle, 
and thus differing interpretations of what to do when you see it in a 
document. Feature or bug? You decide.

At this moment, I'm inclined to the concrete definition (lack of user 
input), and have it be associated with the device, not the service. I 
know others have different opinions. Lets sort this out.

[this is the final open issue I am aware of]

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From simple-bounces@ietf.org  Tue Sep 28 18:47:47 2004
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Subject: Re: [Simple] Pres Data Model Open Issue #3, UUID URN or something else
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Jonathan Rosenberg wrote:
> One of the things the design team spent a lot of time discussing was 
> what a device ID looked like. Our conclusion, as documented in the I-D, 
> is that its a URN. We can't lock down the specific type of URN, but for 
> things where MAC address is a useful unique device identifier, the UUID 
> URN (as it makes use of a MAC address) is a good choice.
> 
> However, upon a more detailed read of the UUID URN specification, I am 
> not sure we can use it. The reason is that the UUID URN spec 
> (http://www.ietf.org/internet-drafts/draft-mealling-uuid-urn-03.txt) 
> describes several ways of computing the URN. There are actually four 
> versions. The time-based version uses the host MAC, along with the 
> current time, to generate the UUID. The name-based version doesnt use 
> time, but relies on a unique namespace, such as DNS or URIs, as a seed. 
> MAC address is not a choice.
> 
> So, the problem is, the UUID form that utilizes the MAC address also 
> includes a time-based component. This serves to provide uniqueness, but 
> makes it useless as a meaningful identifier that other presence sources 
> - like an ethernet switch or 802.11 AP, would be able to provide 
> information about.
> 
> So, the options as I see them are:
> 
> 1. use UUID URN anyway, lose the correlation properties we wanted
> 2. dont specify the URN form at all
> 3. don't specify the URN form in this document, but work on this problem 
> in downstream specifications
> 
> I would propose approach (3). In particular, I think the data model 
> should revert to being agnostic on the URN scheme, reference the UUID 
> URN as one possibility, document its problems. We should develop a 
> separate specification that defines guidelines for choosing the URN for 
> various devices in order to get the correlation properties we want. I 
> also think we should take a stab at defining a few URN schemes that 
> would be useful - MAC and ESN in particular.

I think it does make sense to leave the particular choice out of this 
document.

I do think the problem with UUID can be dealt with in a couple of ways:

- for some kinds of "devices", like softphones and IM clients, it is
   probably fine as-is. They can typically have multiple instances per
   device, so they must construct and instance id and save it
   persistently as part of the instance configuration.

- for a dedicated device, like a phone, that can't or doesn't want to
   use the above technique, a cannonical mac-based uuid can be
   constructed by using a fixed time (e.g. zero) along with the mac.

	Paul


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I agree, although the two examples you give are rather different in 
their usefulness.

In practice, status information for the postal elements is not 
particularly useful as it doesn't tend to change a lot. Your postal 
availability is not exactly something subject to OPEN/CLOSED. Plus, the 
CIPID vcard element already allows to express this.

The in-person indication is definitely more useful for coordination. I'd 
argue you'd actually want a "protocol" indication, not a URN indication, 
since you've already identified the person.

As you said, this can easily be solved by a convention, maybe suitably 
defined in an April 1 RFC, along with the carrier pigeon URL.

Jonathan Rosenberg wrote:

> Currently, there is no easy way to incorporate the notion of "in person" 
> or "written/postal" communications as services represented in the data 
> model. The reason is that there is no URI scheme for them, and no 
> attributes that can be used to define them. The issue is, what to do 
> about this?
> 
> My proposal is to punt the issue in this document. State that it is 
> expected that future specifications will address how to represent these.
> 
> To actually solve the problem, I think the right approach is to define a 
> URN scheme along the lines of "urn:inperson" to represent an in-person 
> communications scheme. For postal communications, I think you want a URN 
> that includes the postal address, something like 
> urn:postal:sname=Lanidex%20Plaza-snumber=600-state=NJ-city=Parsippany-zip=07054 
> or something like that. Thats probably a fairly hard problem to solve, 
> and I think even less important for us to tackle.
> 
> However, we can defer working on either of those in earnest for a bit, 
> until our basic work is done.
> 
> Comments?
> 
> Thanks,
> Jonathan R.

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From simple-bounces@ietf.org  Tue Sep 28 20:42:20 2004
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Subject: RE: [Simple] Review of draft-ietf-simple-message-sessions-08 -Chunking
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From: "Orit Levin" <oritl@microsoft.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <hisham.khartabil@nokia.com>
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Sorry for jumping in late.
This exact topic was among the issues that I sent to the MSRP authors
two weeks ago.

I completely agree with Paul's summary below and that the statement in
Section 6.3.1 saying "it is possible the body is shorter than the
range-end field indicates. This can occur if the sender interrupted a
SEND request unexpectedly" must go away.

BTW Just to ensure that my understanding is correct. In Paul's summary
below the "sender" can be an MSRP relay, is it correct?
An MSRP relay MAY convert a uninterruptible chunk into a interruptible
chunk, but the opposite is not true.

Can we confirm that we have an agreement on the whole issue?

Thanks a lot,
Orit.

> -----Original Message-----
> From: simple-bounces@ietf.org=20
> [mailto:simple-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Monday, September 27, 2004 8:41 AM
> To: hisham.khartabil@nokia.com
> Cc: simple@ietf.org
> Subject: Re: [Simple] Review of=20
> draft-ietf-simple-message-sessions-08 -Chunking
>=20
>=20
>=20
> hisham.khartabil@nokia.com wrote:
> >=20
> >>>>Section 6.3.1 says: "it is possible the body is shorter than the=20
> >>>>range-end field indicates. This can occur if the sender
> >>>
> >>interrupted a
> >>
> >>>>SEND request unexpectedly." Based on what I have=20
> interpretted above,=20
> >>>>this is no longer possible. (To be interrupted it must have had a=20
> >>>>range-end of "*".)
> >>>
> >>>
> >>>I think it is still possible, but it means that it was an
> >>
> >>abnormal behaviour. Perhaps we should include some text=20
> discussing a=20
> >>recovery mechanism for this case.
> >>
> >>It is abnormal in the sense that it can only arise if there is a=20
> >>violation of the specification. Normally we don't specify=20
> corrective=20
> >>action in those cases.
> >=20
> >=20
> > No, I meant device or user behaviour that causes the=20
> sending to be immediately aborted. Nothing to do with=20
> protocol violation. (eg: abnorla termination of the application).
>=20
> As best I can tell, we are saying that you must decide, when=20
> beginning to send, whether to use an interruptible chunk, or=20
> an uninterruptible chunk. If you use an uninterruptible chunk=20
> you specify the range-end as a number and include that many=20
> bytes of body. If you choose an interruptible chunk, you use=20
> "*" for range-end and then you put as much body as you like.
>=20
> An application that chooses to use an uninterruptible chunk,=20
> and then fails to include the full number of bytes for it has=20
> violated the protocol.
>=20
> Effectively, an application should only use an=20
> uninterruptible chunk if it already has all the bytes in hand=20
> and ready to send.
>=20
> 	Paul
>=20
>=20
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>=20

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Markus,

You make a good point about another use case for the pidf manipulation.

Based on this, I think the pidf-manipulation draft needs to more 
carefully outline the use cases. I think it would be good to point out 
that it is ideal for:

(1) manipulating person information,
(2) manipulating services that operate in the absence of agents that 
actively run on behalf of the user (email, for example)

In essence, then, the distinction is that PUBLISH is used when there is 
a software agent of some sort representing a device or service, and the 
ability to use that service or that device is predicated on the 
existence and correct operation of that agent. For the above two cases, 
there need not be a software agent that runs in order for the data to be 
meaningful, and thus xcap-pidf.

Probably it needs to be expressed better, but hopefully the idea makes 
sense.

Thanks,
Jonathan R.

Markus.Isomaki@nokia.com wrote:
> Hi all,
> 
> The approach presented in Data Model for Presence (http://www.ietf.org/internet-drafts/draft-rosenberg-simple-presence-data-model-00.txt) has been in principle approved in SIMPLE WG, but there are still some open issues on details. One of them concerns the scope of XCAP application usage for PIDF manipulation (http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-pidf-manipulation-usage-01.txt). The Data Model draft has the following text in Chapter 10: 
> 
>       One of the source of confusion around the XCAP manipulation of
>       PIDF [17] is that it was unclear as to why one would use it as
>       opposed to PUBLISH.  The presence data model presented here sheds
>       some light on that.  PUBLISH is appropriate for communicating
>       information about services and devices.  PUBLISH is designed for
>       the model where there can be more than one such source (as there
>       is for devices and servcies), and where such state is soft-state
>       (as it should be for devices and services).  However, presentity
>       state is not clearly soft-state, and the model here clearly
>       indicates that each presence document can have a single presentity
>       element.  Thus, it might make sense to change the
>       pidf-manipulation spec to only allow setting of presentity tuples.
>       Now, that doesnt forbid a source from trying to PUBLISH presentity
>       information, but there is clearly a need for a hard-state approach
>       for setting presentity information.
> 
> The concrete issue here is whether XCAP PIDF manipulation should be restricted to setting only presentity tuple, i.e. not allow the setting of service tuples or device information.
> 
> My opinion is that we should not make this kind of restriction. 
> 
> There is at least one use case where "hard state" service tuples make a lot of sense, and that is service tuples for "persistent" services such as e-mail, SMS, MMS or voicemail. These communication means are available for the sender even if the recipient is not on-line wrt. that service, since there is a network-based "inbox" taking care of them. XCAP PIDF manipulation is a better way to describe this kind "off-line" state for such services rather than SIP PUBLISH. (BTW, I believe we should define some new status attribute to this kind of services to clearly indicate the difference between "MMS available but no device on-line" vs. "MMS available with a device on-line" etc. Or probably the standardization fora responsible for e.g. MMS can do that.)
> 
> What might be a useful specification is to say that in a typical scenario information derived from SIP PUBLISH takes presedence over the information derived from XCAP. But that is already in the territory of specifying the composer policy, which I think is a separate effort from the data model of XCAP PIDF manipulation drafts.
> 
> So my proposal is basically to keep the XCAP PIDF manipulation draft as it is at the moment, and not make any restricting statements about its use in the Data Model either. Are there other opinions on this topic?
> 
> Regards,
> 	Markus
>  
> 
> _______________________________________________
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-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: [Simple] Presence Data Model: Identifying services
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Hi Markus,

I think the service-features draft is 50% right. As you also point out,
the URI scheme provides rudimentary service classification that in most
cases is sufficient and provides helpful information for the watcher.

But I also don't quite agree with the way service-features uses prescaps
to advertize service capabilities. A super-UA may boast a range of
capabilities leaving the watcher clueless about the actual combinations
of those capabilities that will work. In fact, the more capabilities the
UA boasts, the more useless the information -- the watcher will have to
resort to "trial-and-error", which is anyway the default behavior in the
absence of any prescaps.

Splitting the information into separate tuples, each describing a
specific "service" offered will suffer from the same or worse
"combinatorial explosion" that the draft describes.

I guess I also somewhat agree with Paul's argument that your proposal
merely adds syntax without assigning semantics. Also, XML already has a
fairly clean namespace structure, so the Java-style naming convention
seems superfluous.

In general, there seems at least three orthogonal things a presentity
may want to communicate to the watcher:

i) what communications mean to use?
	* Including what capabilities this medium has
	* In practice, the <contact> + prescaps 
ii) what "application" am I running?
	* This is basically the service-id element
iii) in what state is that application? 
	* Currently not covered
	* Application specific

This in mind, I think we are ultimately talking about extending the
<status> element to express application-specific statuses. After all, I
may have several SIP-based applications that are all in a different
state. For example, I may be in "accepting-new-chess-moves" for my chess
application (running on IM), but in a "please-don't-disturb" mode for
push to talk.

A status extension would implicitly also carry something akin to a
service-id, but instead of just a simple ID, it would have real
meaningful status information as well.

But, I don't know if there is much to standardize here beyond explicitly
mentioning that this is the best extension point for application-id type
features in presence. Seems natural that we should discuss this point in
the data model for presence.

Cheers,
Aki
  

On Thu, 2004-09-23 at 11:10, ext Markus.Isomaki@nokia.com wrote: 
> Hi all,
> 
> I have a comment on Presence Data Model draft about the identification/characterization of services. (I know this has been discussed for a while already, but I still think the current way is somewhat unadequate.)
> 
> The current Data Model draft says:
> 
>    In this model, services are not explicitly enumerated.  That is,
>    there is no "service" attribute with values of "ptt" or "telephony".
>    Rather, the service is identified in one of two ways.  In many cases,
>    the URI scheme is a clear indicator of service.  An "sms" URI clearly
>    indicates SMS, and a "tel" URI clearly indicates telephony.  For some
>    URIs, there may be many services available, for example, SIP.  For
>    those services, each service has a set of characteristics, each of
>    which has a well-defined meaning, such that a system can
>    unequivocally determine whether or not the service has that
>    characteristic.  This is discussed in more detail in [5]. 
> 
> I agree that the contact URI scheme is the most important piece of information to distinguish what is the "service". (There are some gaps here too, since e.g. the URI schemes for SMS or MMS do not even exist, and there may be other non-IETF protocols that face the same problem.) However, I'm not convinced that listing the signaling/media characteristics of the end-point or service really gives enough information to the watcher to really determine if it can communicate with the advertised service/application. 
> 
> The refenced draft [5] has a following example:
> 
> 5.6 Walkie-talkie
> 
>    The walkie-talkie service allows real-time voice communication
>    between participants.  Only one participant can speak at a time; that
>    is, communication is half-duplex.  Typically, participants press a
>    button to indicate that they are ready to speak, although other
>    mechanism (e.g.  voice activation) are occasionally used.
> 
>    Support for the Walkie-Talkie service can be deduced by observing the
>    presence of a contact URI with a scheme of "sip:", associated with
>    the following minimal set of capabilities: <audio>true</audio>
>    <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
> 
> Presumably this is the same as the service sometimes called Push-to-Talk (PTT, PoC). The concrete problem is that there are several _non-interworking_ variants of this service, that still all would match the characteristics listed above, as the _SIP_ part might still be standard, at least for some methods etc. But still the communication would fail even if the tuple looks OK. (Also if I saw the characteristics above, I could as well determine that they describe normal VoIP application running in (very) some old PC with half-duplex audio drivers, so I claim the service description is hard to make unambiguous in the first place.)  
>   
> I think the main point of a service tuple is to give the watcher enough information to know whether he can really communicate & interoperate with the advertised service. Given that many services taht are using SIP also have propritary or non-SIP features, I don't think the current approach is enough. 
> 
> Of course each of these proprietary services are allowed to define additional status or tuple-level extensions to PIDF. However, I would like to have some kind of "service identifier" as part of the basic framework so that this could be done in a consistent manner. I think it would help especially in making of authorization and composition rules more simple. The current "class" attribute is way too loosely defined.
> 
> So the concrete proposal is to include in RPID a "service id" element that would have a vendor-specific namespace, similar to e.g. vendor-specific XCAP AUIDs. So for instance the SIP-based PTT app from vendor xyz.com would have, e.g. 
> 
> <service-id>com.xyz.ptt</service-id>
> 
> while the PTT app compliant with OMA would have, e.g.
> 
> <service-id>com.openmobilealliance.ptt</service-id>
> 
> in _addition_ to describing sip:, half-duplex audio, and the supported methods. Now:
> 1.) Watcher having one of those apps could see right away whether it is possible to communicate
> 2.) Composer getting PUBLISHes from 2 sources could immediately know that they are from a similar app
> 3.) The app could set its authorization rules using the unique id.
> 
> The main downside I can see in this approach that if there are two different proprietary apps that indeed could communicate at least partially, the watcher might make a conclusion that communication is not possible baed on different service-id. However, in this case the watcher still would have the characteristics visible and could determine that some interworking could be achieved. For this kind of reasons there should be careful text about when to use service-id in the first place, and what can be determined from it.
> 
> Does someone think that this is an absolutely bad idea? If so, how would you envision solving the issues discussed above?
> 
> Regards,
> 	Markus 
> 
> 
> _______________________________________________
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> Simple@ietf.org
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, Simple WG <simple@ietf.org>
From: Mike Hammer <mhammer@cisco.com>
Subject: Re: [Simple] Pres Data Model Open Issue #5: what does idle
  mean?
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Jonathan,

I'm having a sense of deja vu here.  I thought previous discussions led to 
the conclusion that idle means exactly that, i.e. no activity since some 
time point, which may or may not be provided, depending on whether the 
idler felt that this would violate privacy.

The conclusion that one could jump to based on such an indication would be 
implicit to the type of service, device, or person.  The likelihood of a no 
answer is conditional and thus requires some intelligence to guess that.  I 
thought this would be left to humans that have such cognitive skills.

While lack of user input is the most likely basis, this can be fooled or 
not depending on what the thing reporting is.  Does a user include 
automatons?  I would say that is an implementation detail and just leave it 
at a "no activity since" definition.  I see no reason to constrain 
this.  My 2 cents from the peanut gallery.

Mike




At 05:29 PM 9/28/2004 -0400, Jonathan Rosenberg wrote:
>This isn't really an issue with the data model so much as rpid, but its 
>used as an example in the data model.
>
>The question is, in the context of the data model, what does <idle> mean? 
>Does <idle> apply to a person? To a device? To a service? For each case, 
>what does it mean? Does it merely mean, "the likelihood is that attempts 
>to reach this service will result in a no-answer", or is it something more 
>specific, such as "idle indicates that no user input has been provided to 
>this device (service) recently".
>
>The first question is to determine its definition, I think. Once you know 
>what it means, you can decide whether it applies to devices, services or 
>person. I'm somewhat on the fence here. One argument is that you want a 
>concise and measurable definition, so that a PUA can concretely say, "this 
>service is idle", and have the recipient of the document know what it 
>means. That would argue for something based on user input. The problem 
>with that definition is that ultimately, the lack of user input is useful 
>because it is an indicator of the real quantity of interest - the 
>likelihood that I'll get a "no answer". So, why not define it as the data 
>we really want. The problem is that you will see vast differences across 
>devices in declaring themselves idle, and thus differing interpretations 
>of what to do when you see it in a document. Feature or bug? You decide.
>
>At this moment, I'm inclined to the concrete definition (lack of user 
>input), and have it be associated with the device, not the service. I know 
>others have different opinions. Lets sort this out.
>
>[this is the final open issue I am aware of]
>
>Thanks,
>Jonathan R.
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>
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Jonathan,

I agree with you and Markus on the validity of Markus's suggested usage. 
And I agree the kind of clarification you suggest would be helpful. But 
I don't think this is normative behavior, so lets make sure the language 
isn't too strong. (All of these attributes should be possible to 
manipulate *either* via XCAP or PUBLISH.

	Paul

Jonathan Rosenberg wrote:
> Markus,
> 
> You make a good point about another use case for the pidf manipulation.
> 
> Based on this, I think the pidf-manipulation draft needs to more 
> carefully outline the use cases. I think it would be good to point out 
> that it is ideal for:
> 
> (1) manipulating person information,
> (2) manipulating services that operate in the absence of agents that 
> actively run on behalf of the user (email, for example)
> 
> In essence, then, the distinction is that PUBLISH is used when there is 
> a software agent of some sort representing a device or service, and the 
> ability to use that service or that device is predicated on the 
> existence and correct operation of that agent. For the above two cases, 
> there need not be a software agent that runs in order for the data to be 
> meaningful, and thus xcap-pidf.
> 
> Probably it needs to be expressed better, but hopefully the idea makes 
> sense.
> 
> Thanks,
> Jonathan R.
> 
> Markus.Isomaki@nokia.com wrote:
> 
>> Hi all,
>>
>> The approach presented in Data Model for Presence 
>> (http://www.ietf.org/internet-drafts/draft-rosenberg-simple-presence-data-model-00.txt) 
>> has been in principle approved in SIMPLE WG, but there are still some 
>> open issues on details. One of them concerns the scope of XCAP 
>> application usage for PIDF manipulation 
>> (http://www.ietf.org/internet-drafts/draft-ietf-simple-xcap-pidf-manipulation-usage-01.txt). 
>> The Data Model draft has the following text in Chapter 10:
>>       One of the source of confusion around the XCAP manipulation of
>>       PIDF [17] is that it was unclear as to why one would use it as
>>       opposed to PUBLISH.  The presence data model presented here sheds
>>       some light on that.  PUBLISH is appropriate for communicating
>>       information about services and devices.  PUBLISH is designed for
>>       the model where there can be more than one such source (as there
>>       is for devices and servcies), and where such state is soft-state
>>       (as it should be for devices and services).  However, presentity
>>       state is not clearly soft-state, and the model here clearly
>>       indicates that each presence document can have a single presentity
>>       element.  Thus, it might make sense to change the
>>       pidf-manipulation spec to only allow setting of presentity tuples.
>>       Now, that doesnt forbid a source from trying to PUBLISH presentity
>>       information, but there is clearly a need for a hard-state approach
>>       for setting presentity information.
>>
>> The concrete issue here is whether XCAP PIDF manipulation should be 
>> restricted to setting only presentity tuple, i.e. not allow the 
>> setting of service tuples or device information.
>>
>> My opinion is that we should not make this kind of restriction.
>> There is at least one use case where "hard state" service tuples make 
>> a lot of sense, and that is service tuples for "persistent" services 
>> such as e-mail, SMS, MMS or voicemail. These communication means are 
>> available for the sender even if the recipient is not on-line wrt. 
>> that service, since there is a network-based "inbox" taking care of 
>> them. XCAP PIDF manipulation is a better way to describe this kind 
>> "off-line" state for such services rather than SIP PUBLISH. (BTW, I 
>> believe we should define some new status attribute to this kind of 
>> services to clearly indicate the difference between "MMS available but 
>> no device on-line" vs. "MMS available with a device on-line" etc. Or 
>> probably the standardization fora responsible for e.g. MMS can do that.)
>>
>> What might be a useful specification is to say that in a typical 
>> scenario information derived from SIP PUBLISH takes presedence over 
>> the information derived from XCAP. But that is already in the 
>> territory of specifying the composer policy, which I think is a 
>> separate effort from the data model of XCAP PIDF manipulation drafts.
>>
>> So my proposal is basically to keep the XCAP PIDF manipulation draft 
>> as it is at the moment, and not make any restricting statements about 
>> its use in the Data Model either. Are there other opinions on this topic?
>>
>> Regards,
>>     Markus
>>  
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>>
> 


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Subject: Re: [Simple] Pres Data Model Open Issue #1: in-person communications
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The location information is not the presentity, but an attribute of the 
in-person capability.

What about providing a postal availability to tell your friends that you 
are in town, in a hotel, in the lobby for direct communication, rather that 
have two people on opposite sides of the same lobby (perhaps obscured by a 
plant) trying to talk over cell phones?

I think the possibility of using presence to determine proximity is a 
useful indication.  So long as this can be deduced from the information 
provided in the presence document, then this issue is covered.

Mike


At 08:00 PM 9/28/2004 -0400, Henning Schulzrinne wrote:
>I agree, although the two examples you give are rather different in their 
>usefulness.
>
>In practice, status information for the postal elements is not 
>particularly useful as it doesn't tend to change a lot. Your postal 
>availability is not exactly something subject to OPEN/CLOSED. Plus, the 
>CIPID vcard element already allows to express this.
>
>The in-person indication is definitely more useful for coordination. I'd 
>argue you'd actually want a "protocol" indication, not a URN indication, 
>since you've already identified the person.
>
>As you said, this can easily be solved by a convention, maybe suitably 
>defined in an April 1 RFC, along with the carrier pigeon URL.
>
>Jonathan Rosenberg wrote:
>
>>Currently, there is no easy way to incorporate the notion of "in person" 
>>or "written/postal" communications as services represented in the data 
>>model. The reason is that there is no URI scheme for them, and no 
>>attributes that can be used to define them. The issue is, what to do 
>>about this?
>>My proposal is to punt the issue in this document. State that it is 
>>expected that future specifications will address how to represent these.
>>To actually solve the problem, I think the right approach is to define a 
>>URN scheme along the lines of "urn:inperson" to represent an in-person 
>>communications scheme. For postal communications, I think you want a URN 
>>that includes the postal address, something like 
>>urn:postal:sname=Lanidex%20Plaza-snumber=600-state=NJ-city=Parsippany-zip=07054 
>>or something like that. Thats probably a fairly hard problem to solve, 
>>and I think even less important for us to tackle.
>>However, we can defer working on either of those in earnest for a bit, 
>>until our basic work is done.
>>Comments?
>>Thanks,
>>Jonathan R.
>
>_______________________________________________
>Simple mailing list
>Simple@ietf.org
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To: Mike Hammer <mhammer@cisco.com>
Subject: Re: [Simple] Pres Data Model Open Issue #1: in-person communications
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I'm not sure I understand your comment completely, but general physical 
location for presentities, in both civic (street/city) and geospatial 
format, are available in the PIDF-LO document, recently finished. Note 
that postal addresses are often not a good approximation to physical 
location. (You probably don't want to meet at my PO box, but there are 
other cases that are less obvious.)

Mike Hammer wrote:

> The location information is not the presentity, but an attribute of the 
> in-person capability.
> 
> What about providing a postal availability to tell your friends that you 
> are in town, in a hotel, in the lobby for direct communication, rather 
> that have two people on opposite sides of the same lobby (perhaps 
> obscured by a plant) trying to talk over cell phones?
> 
> I think the possibility of using presence to determine proximity is a 
> useful indication.  So long as this can be deduced from the information 
> provided in the presence document, then this issue is covered.
> 
> Mike
> 
> 
> At 08:00 PM 9/28/2004 -0400, Henning Schulzrinne wrote:
> 
>> I agree, although the two examples you give are rather different in 
>> their usefulness.
>>
>> In practice, status information for the postal elements is not 
>> particularly useful as it doesn't tend to change a lot. Your postal 
>> availability is not exactly something subject to OPEN/CLOSED. Plus, 
>> the CIPID vcard element already allows to express this.
>>
>> The in-person indication is definitely more useful for coordination. 
>> I'd argue you'd actually want a "protocol" indication, not a URN 
>> indication, since you've already identified the person.
>>
>> As you said, this can easily be solved by a convention, maybe suitably 
>> defined in an April 1 RFC, along with the carrier pigeon URL.
>>
>> Jonathan Rosenberg wrote:
>>
>>> Currently, there is no easy way to incorporate the notion of "in 
>>> person" or "written/postal" communications as services represented in 
>>> the data model. The reason is that there is no URI scheme for them, 
>>> and no attributes that can be used to define them. The issue is, what 
>>> to do about this?
>>> My proposal is to punt the issue in this document. State that it is 
>>> expected that future specifications will address how to represent these.
>>> To actually solve the problem, I think the right approach is to 
>>> define a URN scheme along the lines of "urn:inperson" to represent an 
>>> in-person communications scheme. For postal communications, I think 
>>> you want a URN that includes the postal address, something like 
>>> urn:postal:sname=Lanidex%20Plaza-snumber=600-state=NJ-city=Parsippany-zip=07054 
>>> or something like that. Thats probably a fairly hard problem to 
>>> solve, and I think even less important for us to tackle.
>>> However, we can defer working on either of those in earnest for a 
>>> bit, until our basic work is done.
>>> Comments?
>>> Thanks,
>>> Jonathan R.
>>
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple

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Aki,

I think you make some good observations here. More inline.

	Paul

Aki Niemi wrote:
> Hi Markus,
> 
> I think the service-features draft is 50% right. As you also point out,
> the URI scheme provides rudimentary service classification that in most
> cases is sufficient and provides helpful information for the watcher.
> 
> But I also don't quite agree with the way service-features uses prescaps
> to advertize service capabilities. A super-UA may boast a range of
> capabilities leaving the watcher clueless about the actual combinations
> of those capabilities that will work.

Well, ideally the capabilities are orthogonal to one another, so that 
all combinations are possible and meaningful. Certainly callee 
capabilities notation that prescaps is based on assumes that the 
capabilities are orthogonal. If this is not the case then some other 
representation is called for. (E.g. multiple tuples)

 > In fact, the more capabilities the
> UA boasts, the more useless the information -- the watcher will have to
> resort to "trial-and-error", which is anyway the default behavior in the
> absence of any prescaps.
> 
> Splitting the information into separate tuples, each describing a
> specific "service" offered will suffer from the same or worse
> "combinatorial explosion" that the draft describes.

To some extent this is in the hands of the device implementors. If there 
are lots of complex rules about what can be used with what, then the 
description of the device with inherently either be complex or imprecise.

The obvious answer is: if it hurts, don't do it.

> I guess I also somewhat agree with Paul's argument that your proposal
> merely adds syntax without assigning semantics. Also, XML already has a
> fairly clean namespace structure, so the Java-style naming convention
> seems superfluous.
> 
> In general, there seems at least three orthogonal things a presentity
> may want to communicate to the watcher:
> 
> i) what communications mean to use?
> 	* Including what capabilities this medium has
> 	* In practice, the <contact> + prescaps 
> ii) what "application" am I running?
> 	* This is basically the service-id element

I finally start to warm to the concept when it is described this way.

I think we already have *some* notion of this concept of application 
within callerprefs/calleecaps: certainly actor=msg-taker (and maybe 
actor=attendant) are describing a form of application.

Perhaps this should be generalized further. If so, I think it has as 
much value for callerprefs as it does for presence, so I would argue for 
adding it to callee caps first and then bringing it into presence via 
prescaps.

I think it is going to be quite difficult to come up with a clear and 
concise discrimination between what is an application and what is 
communications means. Luckily, if done via prescaps there is no real 
need to make the distinction.

> iii) in what state is that application? 
> 	* Currently not covered
> 	* Application specific
> 
> This in mind, I think we are ultimately talking about extending the
> <status> element to express application-specific statuses. After all, I
> may have several SIP-based applications that are all in a different
> state. For example, I may be in "accepting-new-chess-moves" for my chess
> application (running on IM), but in a "please-don't-disturb" mode for
> push to talk.

That seems like kind of a mixed bag for a single tuple. It sounds like a 
valid application for multiple tuples.

(Would I really have have the IM at this address dedicated to chess, 
while the PTT isn't? I don't think so.)

I have a sense that the "applications" that share an address are really 
just part of the means of communication. A "real" application would 
probably utilize all the communication modalities that are available at 
that address - if it didn't, they wouldn't be available.

> A status extension would implicitly also carry something akin to a
> service-id, but instead of just a simple ID, it would have real
> meaningful status information as well.
> 
> But, I don't know if there is much to standardize here beyond explicitly
> mentioning that this is the best extension point for application-id type
> features in presence. Seems natural that we should discuss this point in
> the data model for presence.
> 
> Cheers,
> Aki
>   
> 
> On Thu, 2004-09-23 at 11:10, ext Markus.Isomaki@nokia.com wrote: 
> 
>>Hi all,
>>
>>I have a comment on Presence Data Model draft about the identification/characterization of services. (I know this has been discussed for a while already, but I still think the current way is somewhat unadequate.)
>>
>>The current Data Model draft says:
>>
>>   In this model, services are not explicitly enumerated.  That is,
>>   there is no "service" attribute with values of "ptt" or "telephony".
>>   Rather, the service is identified in one of two ways.  In many cases,
>>   the URI scheme is a clear indicator of service.  An "sms" URI clearly
>>   indicates SMS, and a "tel" URI clearly indicates telephony.  For some
>>   URIs, there may be many services available, for example, SIP.  For
>>   those services, each service has a set of characteristics, each of
>>   which has a well-defined meaning, such that a system can
>>   unequivocally determine whether or not the service has that
>>   characteristic.  This is discussed in more detail in [5]. 
>>
>>I agree that the contact URI scheme is the most important piece of information to distinguish what is the "service". (There are some gaps here too, since e.g. the URI schemes for SMS or MMS do not even exist, and there may be other non-IETF protocols that face the same problem.) However, I'm not convinced that listing the signaling/media characteristics of the end-point or service really gives enough information to the watcher to really determine if it can communicate with the advertised service/application. 
>>
>>The refenced draft [5] has a following example:
>>
>>5.6 Walkie-talkie
>>
>>   The walkie-talkie service allows real-time voice communication
>>   between participants.  Only one participant can speak at a time; that
>>   is, communication is half-duplex.  Typically, participants press a
>>   button to indicate that they are ready to speak, although other
>>   mechanism (e.g.  voice activation) are occasionally used.
>>
>>   Support for the Walkie-Talkie service can be deduced by observing the
>>   presence of a contact URI with a scheme of "sip:", associated with
>>   the following minimal set of capabilities: <audio>true</audio>
>>   <duplex>half</duplex> <methods>INVITE,ACK,BYE,CANCEL</methods>
>>
>>Presumably this is the same as the service sometimes called Push-to-Talk (PTT, PoC). The concrete problem is that there are several _non-interworking_ variants of this service, that still all would match the characteristics listed above, as the _SIP_ part might still be standard, at least for some methods etc. But still the communication would fail even if the tuple looks OK. (Also if I saw the characteristics above, I could as well determine that they describe normal VoIP application running in (very) some old PC with half-duplex audio drivers, so I claim the service description is hard to make unambiguous in the first place.)  
>>  
>>I think the main point of a service tuple is to give the watcher enough information to know whether he can really communicate & interoperate with the advertised service. Given that many services taht are using SIP also have propritary or non-SIP features, I don't think the current approach is enough. 
>>
>>Of course each of these proprietary services are allowed to define additional status or tuple-level extensions to PIDF. However, I would like to have some kind of "service identifier" as part of the basic framework so that this could be done in a consistent manner. I think it would help especially in making of authorization and composition rules more simple. The current "class" attribute is way too loosely defined.
>>
>>So the concrete proposal is to include in RPID a "service id" element that would have a vendor-specific namespace, similar to e.g. vendor-specific XCAP AUIDs. So for instance the SIP-based PTT app from vendor xyz.com would have, e.g. 
>>
>><service-id>com.xyz.ptt</service-id>
>>
>>while the PTT app compliant with OMA would have, e.g.
>>
>><service-id>com.openmobilealliance.ptt</service-id>
>>
>>in _addition_ to describing sip:, half-duplex audio, and the supported methods. Now:
>>1.) Watcher having one of those apps could see right away whether it is possible to communicate
>>2.) Composer getting PUBLISHes from 2 sources could immediately know that they are from a similar app
>>3.) The app could set its authorization rules using the unique id.
>>
>>The main downside I can see in this approach that if there are two different proprietary apps that indeed could communicate at least partially, the watcher might make a conclusion that communication is not possible baed on different service-id. However, in this case the watcher still would have the characteristics visible and could determine that some interworking could be achieved. For this kind of reasons there should be careful text about when to use service-id in the first place, and what can be determined from it.
>>
>>Does someone think that this is an absolutely bad idea? If so, how would you envision solving the issues discussed above?
>>
>>Regards,
>>	Markus 
>>
>>
>>_______________________________________________
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> 
> 
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Jonathan Rosenberg wrote:

> The question is, in the context of the data model, what does <idle> 
> mean? Does <idle> apply to a person? To a device? To a service? For each 
> case, what does it mean? Does it merely mean, "the likelihood is that 
> attempts to reach this service will result in a no-answer", or is it 
> something more specific, such as "idle indicates that no user input has 
> been provided to this device (service) recently".

The likelihood of reachability is defined by other things besides the 
lack of user input or similar metrics. Calling "likelihood of 
reachability" <idle> seems a bit odd, although lack of usage is clearly 
a factor in gauging such probability.

The RPID draft tries to state this currently.


> 
> The first question is to determine its definition, I think. Once you 
> know what it means, you can decide whether it applies to devices, 
> services or person. I'm somewhat on the fence here. One argument is that 
> you want a concise and measurable definition, so that a PUA can 
> concretely say, "this service is idle", and have the recipient of the 
> document know what it means. That would argue for something based on 
> user input. The problem with that definition is that ultimately, the 
> lack of user input is useful because it is an indicator of the real 
> quantity of interest - the likelihood that I'll get a "no answer". So, 

But there are lots of reasons for 'no answer'. I might be in a movie 
theater or driving and decide not to answer your call. I might be out of 
coverage area, etc.

In many cases, <idle> is not a terribly good indication of likelihood of 
answer. I make very few calls on my cell phone, i.e., it would be idle 
by any reasonable definition, but it is certainly reachable.

For a person who walks around with an earbud and is exceeding their 
5000-minutes-a-month allotment, idleness may indeed be an indication of 
not being able to be reached.


> why not define it as the data we really want. The problem is that you 
> will see vast differences across devices in declaring themselves idle, 
> and thus differing interpretations of what to do when you see it in a 
> document. Feature or bug? You decide.

I think the general approach in RPID has been to provide reasonably 
well-defined input data which is then interpreted, combining multiple 
pieces of information and maybe historical data, to guide communications 
or qualitative reachability. (For example, a smart watcher may correlate 
various factors into a reachability score, based on previous "experiments".)

> 
> At this moment, I'm inclined to the concrete definition (lack of user 
> input), and have it be associated with the device, not the service. I 
> know others have different opinions. Lets sort this out.

I disagree with the 'device, not service'. I can easily detect non-usage 
for a service and it seems to be as valuable for that service as for 
some piece of plastic.


> 
> [this is the final open issue I am aware of]
> 
> Thanks,
> Jonathan R.

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To: Henning Schulzrinne <hgs@cs.columbia.edu>
From: Mike Hammer <mhammer@cisco.com>
Subject: Re: [Simple] Pres Data Model Open Issue #1: in-person communications
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On the other hand, not all folks carry GPS devices, so the ability 
somewhere to convert a Lat/Long to a street address is useful.

But, to clarify my early point, if the person location information is 
available via presence, then the need to define an in-person communications 
means is probably moot.

Mike

At 11:19 AM 9/29/2004 -0400, Henning Schulzrinne wrote:
>I'm not sure I understand your comment completely, but general physical 
>location for presentities, in both civic (street/city) and geospatial 
>format, are available in the PIDF-LO document, recently finished. Note 
>that postal addresses are often not a good approximation to physical 
>location. (You probably don't want to meet at my PO box, but there are 
>other cases that are less obvious.)
>
>Mike Hammer wrote:
>
>>The location information is not the presentity, but an attribute of the 
>>in-person capability.
>>What about providing a postal availability to tell your friends that you 
>>are in town, in a hotel, in the lobby for direct communication, rather 
>>that have two people on opposite sides of the same lobby (perhaps 
>>obscured by a plant) trying to talk over cell phones?
>>I think the possibility of using presence to determine proximity is a 
>>useful indication.  So long as this can be deduced from the information 
>>provided in the presence document, then this issue is covered.
>>Mike
>>
>>At 08:00 PM 9/28/2004 -0400, Henning Schulzrinne wrote:
>>
>>>I agree, although the two examples you give are rather different in 
>>>their usefulness.
>>>
>>>In practice, status information for the postal elements is not 
>>>particularly useful as it doesn't tend to change a lot. Your postal 
>>>availability is not exactly something subject to OPEN/CLOSED. Plus, the 
>>>CIPID vcard element already allows to express this.
>>>
>>>The in-person indication is definitely more useful for coordination. I'd 
>>>argue you'd actually want a "protocol" indication, not a URN indication, 
>>>since you've already identified the person.
>>>
>>>As you said, this can easily be solved by a convention, maybe suitably 
>>>defined in an April 1 RFC, along with the carrier pigeon URL.
>>>
>>>Jonathan Rosenberg wrote:
>>>
>>>>Currently, there is no easy way to incorporate the notion of "in 
>>>>person" or "written/postal" communications as services represented in 
>>>>the data model. The reason is that there is no URI scheme for them, and 
>>>>no attributes that can be used to define them. The issue is, what to do 
>>>>about this?
>>>>My proposal is to punt the issue in this document. State that it is 
>>>>expected that future specifications will address how to represent these.
>>>>To actually solve the problem, I think the right approach is to define 
>>>>a URN scheme along the lines of "urn:inperson" to represent an 
>>>>in-person communications scheme. For postal communications, I think you 
>>>>want a URN that includes the postal address, something like 
>>>>urn:postal:sname=Lanidex%20Plaza-snumber=600-state=NJ-city=Parsippany-zip=07054 
>>>>or something like that. Thats probably a fairly hard problem to solve, 
>>>>and I think even less important for us to tackle.
>>>>However, we can defer working on either of those in earnest for a bit, 
>>>>until our basic work is done.
>>>>Comments?
>>>>Thanks,
>>>>Jonathan R.
>>>
>>>
>>>_______________________________________________
>>>Simple mailing list
>>>Simple@ietf.org
>>>https://www1.ietf.org/mailman/listinfo/simple
>
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Subject: Re: [Simple] Pres Data Model Issue #4, indicating the source of data
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Similar work will have to be done for geolocation, as this is quite 
important for 911 services to debug when things go wrong. I suspect some 
of the considerations are similar. (You want to know, for example, 
whether the information is native or has been translated and combined in 
some way.)

Also, in general, different pieces of information are likely to have 
different types of sources. For location information, you want to know 
whether this is from GPS or LORAN. Other items will have different 
labeling requirements.

In short: not only out of scope for a general document for getting-done 
reasons, but likely to be highly content-dependent and best handled for 
each type of information.

Jonathan Rosenberg wrote:

> The data model draft spends a fair bit of time talking about conflicting 
> data, and how a presence server can resolve the conflicts. One of the 
> things it talks about is making use of presence meta-data, and in 
> particular, the source of a piece of presence data, as a tool in 
> deciding how to resolve conflicts.
> 
> The question is, do we want to do any work in defining ways of 
> identifying the source of the presence information?
> 
> I am strongly inclined to say "no", and simply leave this as a potential 
> future issue to address. I would modify the data model draft to talk 
> about meta-data, but not say anything about future standardization or 
> any specific specs where such meta-data are defined.
> 
> Comments?
> 
> Thanks,
> Jonathan R.

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As discussed previously, a device may not even have a MAC or may have 
several (if it is a composite or a multi-homed device). Thus, this seems 
inherently difficult. We went through this whole discussion before, I 
believe.

Jonathan Rosenberg wrote:

> One of the things the design team spent a lot of time discussing was 
> what a device ID looked like. Our conclusion, as documented in the I-D, 
> is that its a URN. We can't lock down the specific type of URN, but for 
> things where MAC address is a useful unique device identifier, the UUID 
> URN (as it makes use of a MAC address) is a good choice.
> 
> However, upon a more detailed read of the UUID URN specification, I am 
> not sure we can use it. The reason is that the UUID URN spec 
> (http://www.ietf.org/internet-drafts/draft-mealling-uuid-urn-03.txt) 
> describes several ways of computing the URN. There are actually four 
> versions. The time-based version uses the host MAC, along with the 
> current time, to generate the UUID. The name-based version doesnt use 
> time, but relies on a unique namespace, such as DNS or URIs, as a seed. 
> MAC address is not a choice.
> 
> So, the problem is, the UUID form that utilizes the MAC address also 
> includes a time-based component. This serves to provide uniqueness, but 
> makes it useless as a meaningful identifier that other presence sources 
> - like an ethernet switch or 802.11 AP, would be able to provide 
> information about.
> 
> So, the options as I see them are:
> 
> 1. use UUID URN anyway, lose the correlation properties we wanted
> 2. dont specify the URN form at all
> 3. don't specify the URN form in this document, but work on this problem 
> in downstream specifications
> 
> I would propose approach (3). In particular, I think the data model 
> should revert to being agnostic on the URN scheme, reference the UUID 
> URN as one possibility, document its problems. We should develop a 
> separate specification that defines guidelines for choosing the URN for 
> various devices in order to get the correlation properties we want. I 
> also think we should take a stab at defining a few URN schemes that 
> would be useful - MAC and ESN in particular.
> 
> Comments?
> 
> Thanks,
> Jonathan R.

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Subject: [Simple] I-D ACTION:draft-ietf-simple-presence-data-model-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the SIP for Instant Messaging and Presence Leveraging Extensions Working Group of the IETF.

	Title		: A Data Model for Presence
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-simple-presence-data-model-00.txt
	Pages		: 39
	Date		: 2004-9-29
	
This document defines the underlying data model and data processing
   operations used by Session Initiation Protocol (SIP) for Instant
   Messaging Leveraging Presence Extensions (SIMPLE) presence agents.
   The data model provides guidance on how to map various communications
   systems into presence documents in a consistent fashion.  The data
   processing operations described here include composition, privacy
   filtering, and watcher filtering.

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To: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Simple] Pres Data Model Open Issue #5: what does idle mean?
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I think idle makes sense for devices, services, and maybe people as 
well. If we say idle can only be used for devices we will have problems 
when there are services that are capable of reporting their own idleness 
but are unable to determine device idleness.

	Paul

Henning Schulzrinne wrote:
> Jonathan Rosenberg wrote:
> 
>> The question is, in the context of the data model, what does <idle> 
>> mean? Does <idle> apply to a person? To a device? To a service? For 
>> each case, what does it mean? Does it merely mean, "the likelihood is 
>> that attempts to reach this service will result in a no-answer", or is 
>> it something more specific, such as "idle indicates that no user input 
>> has been provided to this device (service) recently".
> 
> 
> The likelihood of reachability is defined by other things besides the 
> lack of user input or similar metrics. Calling "likelihood of 
> reachability" <idle> seems a bit odd, although lack of usage is clearly 
> a factor in gauging such probability.
> 
> The RPID draft tries to state this currently.
> 
> 
>>
>> The first question is to determine its definition, I think. Once you 
>> know what it means, you can decide whether it applies to devices, 
>> services or person. I'm somewhat on the fence here. One argument is 
>> that you want a concise and measurable definition, so that a PUA can 
>> concretely say, "this service is idle", and have the recipient of the 
>> document know what it means. That would argue for something based on 
>> user input. The problem with that definition is that ultimately, the 
>> lack of user input is useful because it is an indicator of the real 
>> quantity of interest - the likelihood that I'll get a "no answer". So, 
> 
> 
> But there are lots of reasons for 'no answer'. I might be in a movie 
> theater or driving and decide not to answer your call. I might be out of 
> coverage area, etc.
> 
> In many cases, <idle> is not a terribly good indication of likelihood of 
> answer. I make very few calls on my cell phone, i.e., it would be idle 
> by any reasonable definition, but it is certainly reachable.
> 
> For a person who walks around with an earbud and is exceeding their 
> 5000-minutes-a-month allotment, idleness may indeed be an indication of 
> not being able to be reached.
> 
> 
>> why not define it as the data we really want. The problem is that you 
>> will see vast differences across devices in declaring themselves idle, 
>> and thus differing interpretations of what to do when you see it in a 
>> document. Feature or bug? You decide.
> 
> 
> I think the general approach in RPID has been to provide reasonably 
> well-defined input data which is then interpreted, combining multiple 
> pieces of information and maybe historical data, to guide communications 
> or qualitative reachability. (For example, a smart watcher may correlate 
> various factors into a reachability score, based on previous 
> "experiments".)
> 
>>
>> At this moment, I'm inclined to the concrete definition (lack of user 
>> input), and have it be associated with the device, not the service. I 
>> know others have different opinions. Lets sort this out.
> 
> 
> I disagree with the 'device, not service'. I can easily detect non-usage 
> for a service and it seems to be as valuable for that service as for 
> some piece of plastic.
> 
> 
>>
>> [this is the final open issue I am aware of]
>>
>> Thanks,
>> Jonathan R.
> 
> 
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