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Subject: [Simple] Draft minutes SIMPLE meeting IETF52
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Date: Wed, 9 Jan 2002 11:13:04 -0500

Here is a draft of the simple minutes for IETF52.
Please review and submit corrections before Friday 1/11.

RjS

-------------
SIMPLE - SIP for Instant Messaging and Presence Leveraging Extensions
Minutes - 12/12/2001 1300-1500 MDT - IETF52
Chairs: Jon Peterson - jon.peterson@neustar.com
        Robert Sparks - rsparks@dynamicsoft.com

Summary:

Charter and logistics

* An update to the SIMPLE charter has been submitted
  reflecting the SIP/SIPPING split, the proposed SIP
  change process, and the watcherinfo/CPIM mapping
  deliverables. Publication has been tabled pending
  outcome of disussions in SIPPING (see the SIPPING
  minutes)

* MESSAGE has been handed off to SIP - currently 
  scheduled for last call in May. Brian Rosen will
  work with us to move that call to an earlier date

draft-ietf-simple-presence:

* The presence draft will be modified to rely directly
  on sip-events, which will in turn rely on sip-bis 
  instead of 2543. The result will be shorter documents
  with less opportunity for unintentional contradiction.

* We will provide Alerts (notification with indirect content).
  Primary remaining issue: Lifetime of URI in an Alert.
  Both the sip-events and presence drafts will have to change
  to reflect this.

* There is consensus to establish a presence subscription
  dialog based on the first arriving NOTIFY or 2xx response
  to the SUBSCRIBE

* There is consensus to leave the current forking
  text as is. Subscribers SHOULD accept only one
  NOTIFY. If merging is required, the burden lies
  on the subscriber.

draft-ietf-simple-winfo-package:
draft-ietf-simple-winfo-format:

* Proposed recent format changes accepted with
  the additional removal of most-recently-subscribed

* Sip-events and the format draft will reconcile the
  descriptions of reasons for terminating subscriptions

draft-mankin-im-session-guide:

* Hum against taking this on as a SIMPLE work item
  Opinion was that work on the draft should continue
  in the transport area

* SIMPLE group urged to internalize its guidelines

* Applicability of the requirements being placed on OPES
  wrt intermediary discovery and disclosure needs to be
  discussed

draft-rosenberg-simple-im-transport:

* Weakly expressed support. Moderately strong hum against
  proceeding with this proposal.
  - concerns over keeping IMTP and SIP syncronized
  
* Alternate proposal solicited - if none appears before
  early January, work will proceed with IMTP

* Split opinions on whether a proposal must be held to 
  mankin-im-session-guide


--------------------------------------------------------------------- 
Full Notes (thanks to Brian Stucker - bstucker@nortelnetworks.com) 

Charter Updates: 
Have submitted a new charter. 
* Charter resubmitted a couple of weeks ago 
* Now reflects SIP/SIPPING split and proposed SIP change process. 
* Presence 'extension' deliverable now 'event package and related 
protocol mechanisms' 
* Presence publication tabled in SIMPLE, currently considered in 
SIPPING. 
* Dec 01- Presence event package, watcherinfo, CPIM mapping. 
* Jan 02 - Messaging drafts 
Transferred deliverables 
* Draft-ietf-simple-im-01/draft-ietf-sip-message-00 
  - Status (any SIP chairs want to speak to schedule?) 
  - SIP drafts page say May 02 - be nice if that could be moved in a 
bit. 
    * Brian Rosen: Drafts dates are based on logistics problems. We can 
move it up if we can get the draft ready, get reviewers, list comment, 
etc. But we can move it up if there is a need? 
    * Jon: Would be nice if we could have this in the first quarter? 
    * Brian: We'll look into it. 
* Draft-donovan-publish-requirements-01 
  - An actual proposal discussed in SIPPING yesterday. 
  - Clear that SIPPING will only do requirements for publication. 
    * Where will it go from there? SIP WG? 
    * The IETF would like to do the right thing, and not get too rule 
bound. Hope is that the same people will spend some time in SIPPING and 
make sure that the right approach is formed.  The method document has 
to come out of SIP. 
    * SIPPING needs to look at the problem of whether or not we need a 
new method. 
    * Question begs, is presence publication urgently needed in SIMPLE, 
and that needs to be looked into. 

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

Presence Open Issues: 
* Open issues in presence-04 
  - Dependencies and duplication 
    * Currently 
      - Rfc-2543 <- sip events <- presence 
      - Problem is that sip events is duplicating text from bis-05. 
        * Can we simply make sip events dependent on bis, 
          thereby shortening it. 
      - Presence is duplicating text from sip events. 
    * Proposal 
      - Bis <- sip events <- presence 
        * Bis is nearing completion so timing is not going to be 
          much different 
        * Requires cleanup of redundant text across all three documents
          in progress. This will be done. 
  - Alerts 
    * Would be nice to only be notified of change in presence, not 
      actual presence. 
    * Actual presence could then be fetched with HTTP 
    * Why is this useful? 
      - Large presence documents wouldn't get fragmented. 
      - Large presence docs could be fetched when time is good 
        (ie. Not in a call). 
    * Details 
      - SUBSCRIBE would have Accept header indicate support 
      - Presence of application/uri-list 
      - Still have to support application/cpim-pidf+xml 
      - Etc 
    * Issues with Alerts 
      - Broader than presence. Probably belongs in sip-events 
        * Presence would then state that its allowed. 
      - Security issues 
        * Just because SIP can be authenticated, doesn't mean HTTP can. 
          - Transitive trust vs. e2e 
          - Henning brought up concerns on this. 
      - HTTP SIP interactions 
        * Requires a way for SIP server to manipulate content on web server.

          - Henning brought up concerns on this. 
      - Duration of URL validity 
        * Need to specify - presumably subscription duration. 
    * Proposal 
      - Really useful, really simple 
      - Add to sip-events and presence. 

Henning: Similar to web server problem. 
Jonathan: does require an interaction between HTTP and SIP; don't 
have to do it. 
Henning: Must specify something that will work well. 
Brian: Doesn't have to be HTTP, could use other mechanisms. 
Jonathan: Must develop a baseline. 

Call for opinions: 
Sean: I don't see this as a major issue, or a hard thing to do. 
Adam: I'm in support for this, I don't see a problem. The issues 
that Henning brings up are not completely trivial, but they're 
easy to take care of.  Can be added to sip-events, but doesn't 
see a need. 
Rosenberg: Wants a general mechanism the same across all packages 
and to specify behaviors. Need to specify the duration that the 
presence URL is good for. 
Sean/Rohan: Don't go into such detail about the implementation. 
It's unimportant how long the document sticks around. 
Henning: We need to make sure that the semantics are complete and 
equivalent. 
Brian Stucker: Agrees that need to specify amount of time for URL 
to be valid, just as there is one for subscription. Interval 
doesn't need to be specified. 
Sean: There's no guarantee that the semantics are equivalent. 
Adam: This sounds like the argument about how to encode a piece 
of content versus what it means. 
Rosenberg: This is an automata that is processing this URL, so it 
needs to have an idea of how long the URL is good for. 
Jon: Is there any value in leaving this URL open for our 
application? 
Rohan: I don't think that's what Rosenberg is proposing. 
Diana: Likes the consistiency. Does this trickle to general SIP? 
Henning: The semantics that we are arguing about here are buyer 
beware problems. This is a presence document, versus this 
document is not going to be around after a period of time. We 
wouldn't specify that this is a presence document, we're 
specifying other attributes. 
Adam: I don't have a problem with the proposal, but it sounds 
like the semantics of the usage should be in the presence 
document. 
Jonathan: Need to specify what URL means and have well-defined 
semantics to work. 
Henning: sending a JPEG isn't particularly useful. 
Sean: Your implementation has to handle it - doesn't need 
standardization. 
Adam: How is content encoded Vs. what does it mean? 
Jonathan: Wants to define what it means when you return a URL. 
It's important to remember that a person isn't using this URI, 
an automata is.  
Andrew: This is like W3C semantic web problem. 
Jonathan: Disagree - not solving that problem 
Jon: What do we want the Presence URI to mean? 
Rohan: That's not what Jonathan is proposing. 
Henning: 2 issues: what are the time properties and what is the 
content.  In events, this would not be a presence document.  
The semantics being argued are hints to the implementor - these 
are the semantic expectations.   Agree that semantics need to 
be specified. 
Jonathan: Most web browsers accept *.*.  This URI should be 
very specific and have full semantic content, unlike normal web 
request. 
Rosenberg: Presence package would say that this is a one-shot 
document, etc. 

Conclusion: Continue discussion on the list. 

  - NOTIFY/200 Race condition 
    * ASSUME you want to only accept one NOTIFY 
    * Sip-events says you should accept the one that matches 2xx to 
      SUBSCRIBE. 
    * Proposals 
      - Buffer NOTIFYs until 2xx, then 481 all that non-matching ones. 
      - Accept all NOTIFYs, when 2xx arrives, refersh all non-matching. 
      - Accept first NOTIFY 
        * SUB 2xx is ignored 
      - Accept first message 
        * SUB 2xx if its first 
        * NOTIFY it its first 
    * Probably also an issue for sip-events, not presence. 

  - Forking 
    * Definitely the biggest issue. 
    * Some problems are not presence specific 
      - HERFP 
    * Proposal 
      - Keep what's in the document. If they need aggregation, then put 
        a point in the network and leave it be. 
      - If there are multiple PUA, you get partial presence, no 
        guarantees. 
      - Subscribers SHOULD accept only one NOTIFY 
      - Burden is on the subscriber to implement merging if it wants. 
      - Won't generally be needed based on domain requirements and 
        frequently won't happen because HERFP. 

Conclusion: No time for discussion, taken to list. 
---------------------------------------- 
Watcherinfo Open Issues 
  - Events leading to terminated 
    * See previous discussion 
  - State maintenance for pending/waiting 
    * Possible Dos? Sub is authenticated 
    * -01 will talk about setting policy for this 
      - Tradeoff between user experience and state storage 
Stucker: isn't this just a policy issue? Why does this belong in watcher 
info? 
Rosenberg: When you have watcher subs you need to keep state for things 
being watched.  The fact that there's a watcher info on t you need a 
state. Want to specify a caution with regards to potential for attacks. 
The watcherinfo draft defines a model for presence. We shouldn't have this
in sip-events. 
Stucker: Are you going to add when to indicate when subscription is active?

Jonathan: Both have an approved.  Want these to align between 2 documents -
he's only showing error cases where they don't align

Adam: There are 2 other states: Pending and Active.  These are whys for the
failures. 
Jonathan: Should the state machine be in sip-events.  Watcher -info defines
a model; you could have much more states and a more complex state machine.  

  - Elements in data format 
    * Watcherinfo data 
      - Uri of watcher 
      - Status of subscription 
      - Event which triggered notification 
      - Duration time in last state 
      - First-subscribed 
      - Most-recently subscribed 
      - Added: expiration 
      - Removed: Notify-address 
Chris: let's ditch most-recently subscribed, it's not very useful. 
Rosenberg: Okey-dokey. 
Rosenberg: The watcherinfo draft defines a model for presence. We 
shouldn't have this in sip-events. 
  - Status in winfo/sip events proposal on the list. 

Conclusion: No Comments (thus consensus implied) 
--------------------------------------- 
Guidelines for IM Sessions: 
  * Context: 
    - IM payloads are moving away from short-text only, towards 
      * Images 
      * Video clips 
      * Mp3 and other audio data 
      * Etc. 
      * Large populations of users meeting over shared infrastructure 
        (interconnect of AOL, MSN, etc). 
    - Sessions of Messages: 
      * SIP message is the medium. 
    - Transport Background 
      * In order of concern, reliable transport of serious amounts of 
        data raises concerns of: 
        - Congetion collapse 
          * Will congestion of the system invoke protocol actions that 
            increase data sent into the system? (1986 TCP collapse) 
        - Scope of fate-sharing 
          * Can desired end-to-end behavior and properties be achieved 
            and maintained through intermediaries? 
        - Congestion avoidance and shared effects 
          * When J users share a filled or congested path, and adapt, 
            can they get close to each having 1/Jth of capacity? 

Allison Mankin: We have to be that when we do congestion control that TCP 
adapts it's behavior to the congestion on the link. Want to be sure 
that there aren't unanticipated effects from the extremely large amount 
of traffic. 
  * Guidelines Draft 
    - Purpose is to distill transport concerns, with some related 
      architectural and security concerns. 
    - Session model 
      * When data of IM (or session of IM) is carried within headers 
        seen/modified /operated-on by intermediaries 
      * If you took the MESSAGE, without the headers, and threw it onto 
        an end-to-end TCP connection, it wouldn't create any of these
        issues. If you have UDP in the middle, you can run into 
        congestion problems. 
      * If IM clients stop off at the various intermediaries, they get 
        1/N*M bandwidth (N intermediaries, M IM clients) instead 
        of 1/M, which is a significant penalty. It's because of 
        the TCP queuing system. 

Rosenberg: This could happen for a lot of reasons. 
Allison: We don't want congestion collapse, but there are some other 
interesting things going on. 
Rosenberg: If we use TCP then this isn't our problem. 
Allison: No, the answer is that if you use TCP, don't get greedy 
with the number of intermediaries. 
Christian: There is a fundamental problem here, and it's with TCP. 
Allison: I think we still have some value to look into the transport 
dynamic, and moderate the use of intermediaries. Working on another 
paper to fully research this.  This scenario is a worst case. 

    - Multi-Intermediary protocols 
      * Diameter/Apex/Opes/SIP-SIMPLE 
    - Guidelines for Sessions 
      * Signaling MUST be able to negotiate a common underlying 
        transport 
      * MUST not use UDP because of congestion control issues. 
      * Session/transport solution SHOULD support (MUST?) specifying a 
        single transport protocol for entire session path. 
      * MUST not use multiple parallel reliable transport connections. 

David Oran: Why be so harsh, what's wrong with using UDP in a 
congestion safe way? 
Jon (summary): What we're talking about is that this is the session 
model, not setting up the session. 
Allison: Yes, this is a problem with long streams of data. 
   - Guidelines for Sessions 
     * Does design lead to a well-formed session protocol? 
       - Does the design carry the right data over aggregated 
         transport, session membership, data sequencing, 
         identification and so on? 
       - Is it designed right for the security goals? 
       - Draft-iab-opes-00.txt an RFC-t-be from IAB. 
       - Model for some more architectural guidelines: 
         * SHOULD not interpose without consent, by at least one of the 
           parties 
         * MUST support mechanism for discovery of intermediaries 
         * MUST disclose (either in signaling phase or session) the 
           intermediaries 
         * More... need to discuss in email. 

Rosenberg: The intermediaries are not there to modify the content. It's 
just there to deal with NATs, etc.  Such as in SIP. 
Allison: There is an area where the consent issue may be a bit gray. 
It's not a perfect match. 
Jon: I think we're trying to speak as to what the direction should be, 
not where intermediaries should be deployed. If the client is informed 
about what is going on, then it can take action. 
Henning: I would hesitate to draw too close a parallel on that, because 
there are properties in there that make the predictability issue a bit 
harder. There are issues with discovery and finding that we've had for 
some time. The other thing is that content modification (via headers, 
etc) doesn't necessarily match with the consent model in OPES. As long 
as these are transport entities that don't do much, then you can't 
extrapolate for OPES. 
Rosenberg: Why is this not such a problem for email? 
Allison: We're going to have to wrap up, we're running out of time. 
  * Next? 
    - End goal is for SIMPLE WG to internalize and feel ownership of 
      the guidelines 
    - Document should stay individual or be in the set of WG documents? 
Jon: we did make a good try at generalizing these issues. 
Wcom: I believe that the draft is bloated and it confuses signaling and 
transport, so let's get rid of it. 
Jon: That's unfair, we're responding to what this WG has already gotten 
itself into. 
IMPP WG chair: I don't think that this belongs in either working group. 
------------------------- 
IMTP Proposal: 
  * Motivations 
    - Requirements for IM transport 
      * Congest 
      * Reliable 
      * Arbitrary sizes 
      * Framing 
      * Per-im content typing 
      * E2e privacy, integrity, authenticity 
      * Works through nat 
      * Rapid delivery 
      * Lightweight 
      * Parser reuse 
      * Compressible 
      * Support for different content types 
      * Muxing 
      * Logging 
    - The hard configuration (PC in enterprise 1 going through a ent 1 
      proxy to internet, over to proxy in enterprise 2 to a PC in 
      enterprise 2). 
  * How about SIP? 
    - Pros: 
     * Parser reuse 
     * Proxies as intermediaries 
     * NAT traversal easy 
     * Logging, framing, sequencing all done 
    - Cons: 
     * SIP has features that would get in the way 
       - Record-routing 
       - Redirection 
       - Forking 
     * Messages might go over UDP 
     * Performance of Proxy as pure forwarding intermediary is poor. 
  * Idea: Sip Subset 
    - IMTP proper subset of SIP 
      * Remove unneeded features 
        - Forking, redirection, recursion, record-routing and route 
      * Introduce restrictions 
        - Only TCP 
      * Change protocol field to IMTP/1.0 from SIP/2.0 
        - Most proxies won't route 
        - BUT, one-liner to fix that 
        - Want this to be explicit about what protocol is running 
        - Proxy with proper configuration can route IMTP 
      * No forking, no record-routing, no redirect, TCP only 
      * Can use REGISTER to set up IMTP bindings 

Allison: How hard would it be to put in error checking so that if there 
were an inappropriate header were present it could be dealt with. 
Rosenberg: It is a minimal SIP request 
Mic: How do we specify IMTP? Where will this be done, where will 
discussion be held for this since it's a different transport protocol. 
Rosenberg: There will be a separate document for IMTP, and will go into 
this WG. It is burdensome to tie SIP and IMTP together, so let's not. 
Rosenberg: We're working on the end-to-end nature of the protocol to 
specify a key for the session to create an end-to-end solution. We can 
use that key into S/MIME so I can encrypt the content of the message 
path. 

Security:  working on SDP extensions for key exchange (MIKEY) (2 
pass)  Could use key as input to S/MIME to encrypt e2e.   Whatever 
solutions we develop for SIP would apply to IMTP. 

Andrew: What is our reaction? It's not obvious that there's not good 
engineering necessarily here, but this is what we need to do to get 
things done. 
Henry: What is the problem we're trying to solve? We have fast servers, 
we have bandwidth, so why is this a problem. 
Jonathan: Want a session model for SIP.  
Cullen: Can deal with IMTP being a profile in SIP.  Can't deal with 
divergences between the 2.  Should keep synced with SIP. 
Henning: As long as it's done by reference, then it's not as big of a 
problem. 
Rosenberg: It's not that clear, but it's not by reference. 
Dean: Either IMTP reduces to the argument against a session model or 
argues against using SIP how we're using it now.  Believes that if SIP 
as we've defined it doesn't scale then it's broke, but if it's not 
broke, then why can't it be used as is for the session MESSAGE problem.  
Rosenberg: SIP has the idea of finding the user, and the delivery of 
the content. The session model for SIP for SIP works by finding the 
user, then delivering the large data by another piece. SIP is good for 
the discovery problem and NOT good for the delivery of large data. 
Dean: I would argue that if we can't do signaling for call setup, then 
we can't do messaging, and vice versa. If you want to use a new 
protocol, why not use BEEP or another protocol? 

Rosenberg: Nobody has come forth. 
Rohan: I think we should make it so that it absolutely should not go 
through a proxy. 
David Oran: if there is anything to be gained by the MESSAGE method in 
the IMTP versus in SIP then let's converge them. I don't think it's a 
big deal with IMTP being renamed. If people think profiling is a 
reasonable way, then look at MEGACO.  Is there anything to be gained 
from the Message method in paging and the message method in session? 
People have said that it's easier. 
Dave Oran: Could you share a TCP connection with SIP? 
Jonathan: No, but you can share TCP between IMTP. 
Rosenberg: I am getting the feeling that it's not a good consensus. 
Christian: I like this proposal because it's easy to specify. Do we 
just have MESSAGE, nothing else, and enable us to do this? It's easy to 
implement, easy to undertand, etc. 
Henning: We have to solve this problem. 

Chairs consensus gathering: 

* Do we want to pursue this? Hum vote: 
  - Yes? Delayed hum.  
  - No? stronger hum. 

* Is the session model for messaging important? 
  - Yes?  Hum 
  - No? small hum. 

Rohan: Proposes that unless you get alternative proposals within a 
reasonable timeframe, that we should just go with this. 
Jon: We'll  send something out on the list, and if we don't get a 
response by the end of the holidays, we'll go with IMTP by default. 

Chairs consensus gathering: 

* If we are going to have a proposal for Sessions of messages, should 
they adhere to guidelines? 
  - Yes? Hum 
  - No? Equal hum 



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-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Tuesday, January 08, 2002 7:46 AM
Subject: I-D ACTION:draft-mrose-simple-exchange-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: IM Simple Exchange (IMSX)
	Author(s)	: M. Rose
	Filename	: draft-mrose-simple-exchange-00.txt
	Pages		: 27
	Date		: 07-Jan-02
	
The Common Presence and Instant Messaging profile (CPIM) is a set of
syntax and semantics for instant messaging (IM) and online presence
services, independent of the underlying transfer infrastructure.  The
Session Initiation Protocol (SIP) negotiates session information for
media streams.  This memo defines a BEEP profile, IMSX, for
exchanging CPIM messages after SIP has performed signalling.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mrose-simple-exchange-00.txt

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From simple-admin@mailman.dynamicsoft.com  Wed Jan  9 19:17:01 2002
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> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-mrose-simple-exchange-00.txt

if you prefer the html version, go to:

    http://www.beepcore.org/beepcore/docs/profile-imsx.html

enjoy!

/mtr


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From simple-admin@mailman.dynamicsoft.com  Thu Jan 10 12:05:16 2002
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Subject: [Simple] Re: [Sip] simple-presence-04
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Aymeric MOIZARD wrote:

> My question is about the tupple element in
> draft-ietf-simple-presence-04.txt
> (Is it the correct mailing list to use?)


The simple list is the right list. I've changed the recipient line.


> 
> The draft draft-ietf-simple-presence-04.txt shows examples
> of "application/cpim-pidf+xml" body where "name" is an
> attribute of "tuple".
> 
>      <presence xmlns="http://www.ietf.org/ns/cpim-pidf-xml-1.0">
>         <tuple name="ph-1">
>           <status>
>             <value>open</value>
>             <detail type="phone"
>             
> schema="http://www.ietf.org/dtd/im-type-phone.dtd">on-hook</detail>
>           </status>
>           <contact priority="2">sip:user@example.com</contact>
>         </tuple>
>       </presence>
> 
> I found that definition in draft-ietf-impp-cpim-pidf-01.txt
> where "id" is a mandatory attribute for tuple.
> 
> <!ELEMENT tuple (status,contact?,note?,timestamp?)>
> <!ATTLIST tuple
>              id   %TUPLEID;      #REQUIRED
> 
> 
> The example and defintions does not match!
> Did I choose wrong versions of drafts? Any other
> explanations?


The two docuemtns have been evolving idenpdently, and fall out of sync 
as they evolve. When the pidf spec is stable, we can make sure the 
presence doc formats are all up to date.

Thanks,
Jonathan R.




-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com

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From simple-admin@mailman.dynamicsoft.com  Thu Jan 10 14:58:09 2002
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From: Rohan Mahy <rmahy@cisco.com>
To: "Marshall T. Rose" <mrose+mtr.netnews@dbc.mtview.ca.us>
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Subject: Re: [Simple] FW: I-D ACTION:draft-mrose-simple-exchange-00.txt
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Date: Thu, 10 Jan 2002 11:31:25 -0800 (PST)

Hi Marshall,

I have a few suggestions and questions on your draft.

1) In the security considerations section you use MUST strength for both
DIGEST-MD5 and TLS_RSA_WITH_3DES_EDE_CBC_SHA.  I think requiring TLS and
3DES is too restrictive, especially for small devices or lightweight
implementations.  I propose using MUST strength for authentication
and integrity via DIGEST_MD5, and RECOMMENDED or SHOULD strength for
privacy and "all".

2) Please briefly (one sentence) mention the I: and L: convention before
you use it in the examples.

3a) You mention middleboxes briefly, but I wasn't sure what you are really
recommending when you say it may be needed for  "proxies to transparently
rewrite the transport information "   I assume you mean SIP proxies?
This is *AN* approach, but we won't be able to get to end-to-end integrity
if this is required, hence my next question...

3b) I am assuming that IMSX assumes that the two BEEP entities will
communicate directly without relays.  How would this work with a
couple of BEEP relays involved?  What address would you advertise in SDP?
If this is possible, I think this is very useful for getting around the
middlebox traversal problem.

4) I didn't get a good sense from your examples how to come up with
unique cid: URLs (especially if the originator is behind a NAT).
Do these have to be unique for all users/sessions, per BEEP session, or
per beep request?

5) I think an example or call-flow showing multiplexed bidirectional
messages would be really useful.

6) I'm assuming that the proposed transformation between im: and sip: URLs
is to merely replace the scheme.  Is this correct?

thanks,
-rohan



On Wed, 9 Jan 2002, Marshall T. Rose wrote:

> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-mrose-simple-exchange-00.txt
>
> if you prefer the html version, go to:
>
>     http://www.beepcore.org/beepcore/docs/profile-imsx.html
>
> enjoy!
>
> /mtr
>
>
> _______________________________________________
> simple mailing list
> simple@mailman.dynamicsoft.com
> http://mailman.dynamicsoft.com/mailman/listinfo/simple
>

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From simple-admin@mailman.dynamicsoft.com  Thu Jan 10 19:32:32 2002
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Date: Thu, 10 Jan 2002 16:27:17 -0800
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> 1) In the security considerations section you use MUST strength for both
> DIGEST-MD5 and TLS_RSA_WITH_3DES_EDE_CBC_SHA.  I think requiring TLS and
> 3DES is too restrictive, especially for small devices or lightweight
> implementations.  I propose using MUST strength for authentication
> and integrity via DIGEST_MD5, and RECOMMENDED or SHOULD strength for
> privacy and "all".

the particular wording you see, and the particular technologises list
reflect what the IESG is letting go through these days. in other words,
they've been returning drafts to working groups and individual submissions,
asking for language with these choices.

it is important to appreciate the distinction between what gets implemented
and what gets provisioned. the requirement is that tls be available in
implementations, not that it has to be used by administrators.

implementors are free to sell products with other technologies.
administrators are free to configure or not whatever comes with the
software.

so what all this means is that administrators responsible for people with
small devices won't turn privacy on. that's okay. what's important is that,
two administrators agreeing to use privacy know that there will be at least
one choice available, tls, regardless of what products they're using. they
can then make a reasoned cost-benefit analysis as to what gets provisioned.


> 2) Please briefly (one sentence) mention the I: and L: convention before
> you use it in the examples.

ok.


> 3a) You mention middleboxes briefly, but I wasn't sure what you are really
> recommending when you say it may be needed for  "proxies to transparently
> rewrite the transport information "   I assume you mean SIP proxies?
> This is *AN* approach, but we won't be able to get to end-to-end integrity
> if this is required, hence my next question...

>
> 3b) I am assuming that IMSX assumes that the two BEEP entities will
> communicate directly without relays.  How would this work with a
> couple of BEEP relays involved?  What address would you advertise in SDP?
> If this is possible, I think this is very useful for getting around the
> middlebox traversal problem.

i don't think the text could be any clearer: the IMSX peers have no interest
in the network topology. the SIP infrastructure is responsible for ensuring
end-to-end usability of the transport information. if middleboxes are
involved, then the SIP infrastructure will have to deal with this. the
problem is no different for any other application protocol making use of the
SIP infrastructure.

there is no explicit relaying model. if you want that, this is the wrong
architecture. if you want to solve the middlebox problem, this is the wrong
working group.


> 4) I didn't get a good sense from your examples how to come up with
> unique cid: URLs (especially if the originator is behind a NAT).
> Do these have to be unique for all users/sessions, per BEEP session, or
> per beep request?

i think the email folks solved this problem long ago. generating a unique
url is trivial, particularly, since it is used to refer only to data
contained within the message...


> 5) I think an example or call-flow showing multiplexed bidirectional
> messages would be really useful.

well, there are lots of call-flow diagrams in the sip document, to pick one
and i guess i can copy it...


> 6) I'm assuming that the proposed transformation between im: and sip: URLs
> is to merely replace the scheme.  Is this correct?

that's really for the group to decide...

/mtr


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From simple-admin@mailman.dynamicsoft.com  Thu Jan 10 23:11:58 2002
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From: Rohan Mahy <rmahy@cisco.com>
To: "Marshall T. Rose" <mrose+internet.ietf.simple@dbc.mtview.ca.us>
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Date: Thu, 10 Jan 2002 18:08:42 -0800 (PST)

Hi Marshall,

Thanks for the quick response.  A few more comments inline...

On Thu, 10 Jan 2002, Marshall T. Rose wrote:

> > 1) In the security considerations section you use MUST strength for both
> > DIGEST-MD5 and TLS_RSA_WITH_3DES_EDE_CBC_SHA.  I think requiring TLS and
> > 3DES is too restrictive, especially for small devices or lightweight
> > implementations.  I propose using MUST strength for authentication
> > and integrity via DIGEST_MD5, and RECOMMENDED or SHOULD strength for
> > privacy and "all".
>
> the particular wording you see, and the particular technologises list
> reflect what the IESG is letting go through these days. in other words,
> they've been returning drafts to working groups and individual submissions,
> asking for language with these choices.
>
> it is important to appreciate the distinction between what gets implemented
> and what gets provisioned. the requirement is that tls be available in
> implementations, not that it has to be used by administrators.

the problem is that some lightweight SIP implementations (e.g phones) will
be unable to implement BEEP and TLS and RSA and 3DES.  this is not an
administrator level problem, it is an implementation burden.

if I haven't convinced you, then I will talk to the appropriate AD.  This
shouldn't be your problem.

> implementors are free to sell products with other technologies.
> administrators are free to configure or not whatever comes with the
> software.
>
> so what all this means is that administrators responsible for people with
> small devices won't turn privacy on. that's okay. what's important is that,
> two administrators agreeing to use privacy know that there will be at least
> one choice available, tls, regardless of what products they're using. they
> can then make a reasoned cost-benefit analysis as to what gets provisioned.
>
>
> > 2) Please briefly (one sentence) mention the I: and L: convention before
> > you use it in the examples.
>
> ok.
>
>
> > 3a) You mention middleboxes briefly, but I wasn't sure what you are really
> > recommending when you say it may be needed for  "proxies to transparently
> > rewrite the transport information "   I assume you mean SIP proxies?
> > This is *AN* approach, but we won't be able to get to end-to-end integrity
> > if this is required, hence my next question...
>
> >
> > 3b) I am assuming that IMSX assumes that the two BEEP entities will
> > communicate directly without relays.  How would this work with a
> > couple of BEEP relays involved?  What address would you advertise in SDP?
> > If this is possible, I think this is very useful for getting around the
> > middlebox traversal problem.
>
> i don't think the text could be any clearer: the IMSX peers have no interest
> in the network topology. the SIP infrastructure is responsible for ensuring
> end-to-end usability of the transport information. if middleboxes are
> involved, then the SIP infrastructure will have to deal with this. the
> problem is no different for any other application protocol making use of the
> SIP infrastructure.

How about:

"Note that with the introduction of middleboxes [7], such as firewalls and
network address translators, IMSX peers are responsible for using
transport addresses which are usable end-to-end by their peers, or sending
SIP through proxies or other entities which rewrite these addresses as
appropriate to allow end-to-end communication."   ??

> there is no explicit relaying model. if you want that, this is the wrong
> architecture. if you want to solve the middlebox problem, this is the wrong
> working group.

sorry, I was thinking of APEX.

> > 4) I didn't get a good sense from your examples how to come up with
> > unique cid: URLs (especially if the originator is behind a NAT).
> > Do these have to be unique for all users/sessions, per BEEP session, or
> > per beep request?
>
> i think the email folks solved this problem long ago. generating a unique
> url is trivial, particularly, since it is used to refer only to data
> contained within the message...

ok, you may want to mention that the url is unique only within the
message.

thanks again,
-rohan

>
> > 5) I think an example or call-flow showing multiplexed bidirectional
> > messages would be really useful.
>
> well, there are lots of call-flow diagrams in the sip document, to pick one
> and i guess i can copy it...
>
>
> > 6) I'm assuming that the proposed transformation between im: and sip: URLs
> > is to merely replace the scheme.  Is this correct?
>
> that's really for the group to decide...
>
> /mtr
>
>
>

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From simple-admin@mailman.dynamicsoft.com  Fri Jan 11 13:27:43 2002
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Hi,


New IMTP-related draft is now available:
http://www.ietf.org/internet-drafts/draft-koskelainen-imtpext-00.txt

"IMTP extensions for message threading"

Abstract

The IMTP specification defines a session-based IM transport
as well as the message format for it. However, it does not support  
message threads or discussion topics which are useful in 
applications such as multi-party chat. This document defines 
extensions to IMTP which allow applications to utilize threads 
and sub-threads within an IMTP message session. Also, the 
extensions enable message identification.



BR,
--
Petri
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> > it is important to appreciate the distinction between what gets
implemented
> > and what gets provisioned. the requirement is that tls be available in
> > implementations, not that it has to be used by administrators.
>
> the problem is that some lightweight SIP implementations (e.g phones) will
> be unable to implement BEEP and TLS and RSA and 3DES.  this is not an
> administrator level problem, it is an implementation burden.
>
> if I haven't convinced you, then I will talk to the appropriate AD.  This
> shouldn't be your problem.

sure.


> How about:
>
> "Note that with the introduction of middleboxes [7], such as firewalls and
> network address translators, IMSX peers are responsible for using
> transport addresses which are usable end-to-end by their peers, or sending
> SIP through proxies or other entities which rewrite these addresses as
> appropriate to allow end-to-end communication."   ??

i don't think this works because of the "IMSX peers are responsible" part.
this implies some kind of topology-specific behavior on the part of IMSX,
which isn't there. IMSX doesn't know, nor care, about topology. that's
someone else's job... the requirement on the part of IMSX is stated in the
previous paragraph, to wit:

    As with any model based on dynamic rendezvous,
    it is critically important that each application faithfully record the
    appropriate transport information in the corresponding SDP
    announcement.


> > > 4) I didn't get a good sense from your examples how to come up with
> > > unique cid: URLs (especially if the originator is behind a NAT).
> > > Do these have to be unique for all users/sessions, per BEEP session,
or
> > > per beep request?
> >
> > i think the email folks solved this problem long ago. generating a
unique
> > url is trivial, particularly, since it is used to refer only to data
> > contained within the message...
>
> ok, you may want to mention that the url is unique only within the
> message.

ok.

/mtr


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From simple-admin@mailman.dynamicsoft.com  Fri Jan 11 18:06:06 2002
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On Fri, 11 Jan 2002, Marshall T. Rose wrote:
[snip]
> > How about:
> >
> > "Note that with the introduction of middleboxes [7], such as firewalls and
> > network address translators, IMSX peers are responsible for using
> > transport addresses which are usable end-to-end by their peers, or sending
> > SIP through proxies or other entities which rewrite these addresses as
> > appropriate to allow end-to-end communication."   ??
>
> i don't think this works because of the "IMSX peers are responsible" part.
> this implies some kind of topology-specific behavior on the part of IMSX,
> which isn't there. IMSX doesn't know, nor care, about topology. that's
> someone else's job... the requirement on the part of IMSX is stated in the
> previous paragraph, to wit:
>
>     As with any model based on dynamic rendezvous,
>     it is critically important that each application faithfully record the
>     appropriate transport information in the corresponding SDP
>     announcement.

One of the ways that we are getting SIP intiated media through middleboxes
is to have the SIP UA "lie" about its address.  A SIP UA might use STUN to
get public addresses for RTP and RTCP media, or TURN to establish use of a
media relay.  What I meant by IMSX peer was the host that IMSX runs
on.  An alternate rewording would be "SIP User Agents implementing
IMSX are responsible..."   This is an alternative to the requirement
for a proxy to do rewriting.  Some relevant pointers to this method of
middlebox traversal are below.

draft-rosenberg-midcom-stun-00.txt
draft-rosenberg-midcom-turn-00.txt
draft-rosenberg-sipping-nat-scenarios-00.txt

hope this makes sense.
thanks,
-rohan


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> One of the ways that we are getting SIP intiated media through middleboxes
> is to have the SIP UA "lie" about its address.  A SIP UA might use STUN to
> get public addresses for RTP and RTCP media, or TURN to establish use of a
> media relay.  What I meant by IMSX peer was the host that IMSX runs
> on.  An alternate rewording would be "SIP User Agents implementing
> IMSX are responsible..."   This is an alternative to the requirement
> for a proxy to do rewriting.  Some relevant pointers to this method of
> middlebox traversal are below.
>
> draft-rosenberg-midcom-stun-00.txt
> draft-rosenberg-midcom-turn-00.txt
> draft-rosenberg-sipping-nat-scenarios-00.txt
>
> hope this makes sense.

yes, it makes sense, i just question whether it belongs in imsx. how about
this text:

   Note that with the introduction of middleboxes [7], such as firewalls
   and network address translators, it may be necessary to rewrite the
   transport information in order to account for administratively-
   mandated media gateways or address translation.  Accordingly, SIP
   user agents are responsible for either directly rewriting the
   transport information, or, sending the SDP announcement through the
   appropriate intermediaries that will transparently rewrite the
   transport information.  In a phrase, "res ipsa loquitur".

/mtr


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From simple-admin@mailman.dynamicsoft.com  Fri Jan 11 19:21:06 2002
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From: Rohan Mahy <rmahy@cisco.com>
To: "Marshall T. Rose" <mrose+internet.ietf.simple@dbc.mtview.ca.us>
cc: <simple@mailman.dynamicsoft.com>, Marshall Rose <mrose@dbc.mtview.ca.us>
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Date: Fri, 11 Jan 2002 16:18:27 -0800 (PST)



On Fri, 11 Jan 2002, Marshall T. Rose wrote:

>
> > One of the ways that we are getting SIP intiated media through middleboxes
> > is to have the SIP UA "lie" about its address.  A SIP UA might use STUN to
> > get public addresses for RTP and RTCP media, or TURN to establish use of a
> > media relay.  What I meant by IMSX peer was the host that IMSX runs
> > on.  An alternate rewording would be "SIP User Agents implementing
> > IMSX are responsible..."   This is an alternative to the requirement
> > for a proxy to do rewriting.  Some relevant pointers to this method of
> > middlebox traversal are below.
> >
> > draft-rosenberg-midcom-stun-00.txt
> > draft-rosenberg-midcom-turn-00.txt
> > draft-rosenberg-sipping-nat-scenarios-00.txt
> >
> > hope this makes sense.
>
> yes, it makes sense, i just question whether it belongs in imsx. how about
> this text:
>
>    Note that with the introduction of middleboxes [7], such as firewalls
>    and network address translators, it may be necessary to rewrite the
>    transport information in order to account for administratively-
>    mandated media gateways or address translation.  Accordingly, SIP
>    user agents are responsible for either directly rewriting the
>    transport information, or, sending the SDP announcement through the
>    appropriate intermediaries that will transparently rewrite the
>    transport information.  In a phrase, "res ipsa loquitur".
>
> /mtr

sounds good.

thanks,
-rohan

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From simple-admin@mailman.dynamicsoft.com  Tue Jan 22 23:07:49 2002
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From: "Tang Lihua" <tanglihua2043@hotmail.com>
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Subject: [Simple] Somes questions about behavior of PS
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Date: Wed, 23 Jan 2002 11:52:29 +0800

Hi,guys

I want to clarify the behaviour of PS. My questions are as follows: 

About Registrar
1. How to upload buddylist or watcherlist to PS? Through registration or 
other way? If a buddylist is stored in PS, should PS automaticly send out 
subscription to each buddy on the list? And it's strange that in this case 
a watcher will receive notifications but no subscription is sent out by the 
watcher itself. Also another problem is that it will burden PS with 
handling all the buddylists or watcherlists. 
2. If a user gets unregistered, remove this user from database or just mark 
it as 'Unregistered'?
How can the PS know an offline presentity is a once registered user? Is 
User Managerment a need? 
 
About Authentication
1. Is the authentication in PS a server-side(as proxy) or a ua-side one(as 
PA) one? If a subscription is sent to an offline presentity, should the PS 
challenge the subscriber?
2. Should the PS maintain access password lists of each presentity for each 
possible watcher? Is there a better way for PS to authorize the watcher?
2. Draft says, 'For pending subscriptions, the state of the presentity 
SHOULD include some kind of textual note that indicates a pending status.' 
Is there any solution to it till now? More, how can the PS notify the 
watcher that a buddy is offline now or a buddy gets online?

Thanks for your comments!

Best regards,

Tang Lihua

_________________________________________________________________
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From simple-admin@mailman.dynamicsoft.com  Thu Jan 24 00:41:25 2002
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Subject: [Simple] Call-ID & CSeq changes in MESSAGE requests
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Hi All,
I have been reading through the draft on Sip Extensions for Instant 
Messaging (simple-im-01) and I have a doubt regarding the change of 
Call-ID & CSeq headers. Since MESSAGE requests do not create an implied 
session (implying no call-leg, or call-state), does this mean that the 
Call-leg keeps changing and the CSeq MAY remain the same in subsequent 
MESSAGE requests, or vice versa?

Thanks & Regards,
Nirmalya

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From simple-admin@mailman.dynamicsoft.com  Thu Jan 24 10:25:47 2002
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Date: Thu, 24 Jan 2002 10:22:51 -0500
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Nirmalya Ghosh wrote:
> 
> Hi All,
> I have been reading through the draft on Sip Extensions for Instant
> Messaging (simple-im-01) and I have a doubt regarding the change of
> Call-ID & CSeq headers. Since MESSAGE requests do not create an implied
> session (implying no call-leg, or call-state), does this mean that the
> Call-leg keeps changing and the CSeq MAY remain the same in subsequent
> MESSAGE requests, or vice versa?

Yes, the Call-ID will change in each subsequent message. Furthermore,
since CSeq is defined within the space of a call leg, the value of the
CSeq header in each message are really independent and unrelated.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
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From simple-admin@mailman.dynamicsoft.com  Tue Jan 29 13:26:39 2002
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From: =?iso-8859-1?Q?H=E9l=E8ne?= Montarou <montarou@inrs-telecom.uquebec.ca>
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Subject: [Simple] Misunderstanding of Presence Service with SIP
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Date: Tue, 29 Jan 2002 13:25:03 -0500

This is a multi-part message in MIME format.
--------------EA092FC0C53693519E1A26AE
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: 8bit

Dear Sir,

 I am studying SIP and the presence service.
 There is some points I misunderstand concerning the model proposed in
 RFC 2778 and the analogy between PUA's addresses and
 presentities'addresses.

 A. Concerning the model :

 So, if I understand well:
 - A principal can manipulate zero or more presentities,
 - A principal interacts with the system via one or several user agents
 so one or several Presence User Agent (PUA),
 - A PUA can not have several presentities due to an addressing problem
 but several PUAs can send their presence information to the same
 presentity.

 Finally, the configuration in the document attached (configuration.ps)
are allowed.

  Is it correct from now?

 - We supposed all the entities are colocalised.
 In the second case, PUAs share the same address so that the presentity
 can be reach via this same address, is not it ??
 In this case, we could say that the address identify uniquely the
 principal.
 In the first case, presentities have different addresses. Only PUAs of
 the Principal can be identify uniquely through presentities. A watcher
 which would like to know the principal presence information (state of
 all devices and services on devices of this principal) has to know the
 PUAs ' presentities addresses, is not it ??
 As an example , PUAs could be different programs/applications.. so
 services on the same computer like email, phone, conference.. etc.. is
 that correct ?   could you give me some more example ?

 -  When PUAs and presentities are delocalised, they can have different
 addresses.
 If we consider both configuration above, how is it possible to identify

 a PUA uniquely for a watcher ? and principal uniquely ?
 What should be the presentity address in particular in the second case
?

 As an example, PUAs could be programs/applications/several devices
 (mobile phone, fixe phone and PC ) with three different addresses. For
 this example, is the second configuration still allowed?

 B.  Concerning the CPIM format:

 - The presence information contained  in a tuple concern a PUA.
 - A list of tuples for the same presentity means that several PUAs send

 information at the same presentity.

 Is it correct ?

 Here the example from the Internet Draft "CPIM Presence Information
Data
 Format":

  <presence xmlns="urn:ietf:params:cpim-presence:">
      <presentity id="pres:shingo@jp.fujitsu.com"/>
         <tuple id="mobile-im">
             <status>
                 <value>OPEN</value>
                 <value

type="urn:ietf:params:cpim-presence:status-type:im">BUSY</value>
                 <value
                   type="urn:example-com:cpim-status-type:location"

schema="http://www.example.com/impp/location.dtd">HOME</value>
              </status>
              <contact
priority="2">im:shingo@mobilecarrier.ne.jp</contact>
              <note>Don't Disturb Please!</note>
              <timestamp>2001-10-27T16:49:29Z</timestamp>
         </tuple>
         <tuple id="email">
             <status>
             <value>OPEN</value>
             </status>
            <contact priority="1">mailto:shingo@jp.fujitsu.com</contact>

         </tuple>
         <note>I'll be in Tokyo tomorrow</note>
   </presence>

 Could you explain it in terms of Principal, PUA, Presentity, devices,
services and
 addresses ?

 Waiting for your answer, I thank you very much in advance.
 This would help me a lot in my understanding of SIP and presence.

 Regards,

H閘鑞e Montarou.

 ----------------------------------------------
 INRS-T閘閏ommunications
 Place Bonaventure
  900, de la Gaucheti鑢e Ouest
 Niveau C, Case Postale 644
 Montr閍l, QC
 H5A 1C6 Canada

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

--------------EA092FC0C53693519E1A26AE
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%%CreationDate: Mon Jan 28 13:20:51 2002
%%For: montarou@paloverde (Helene Montarou)
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/col20 {0.820 0.000 0.000 srgb} bind def
/col21 {0.560 0.000 0.560 srgb} bind def
/col22 {0.690 0.000 0.690 srgb} bind def
/col23 {0.820 0.000 0.820 srgb} bind def
/col24 {0.500 0.190 0.000 srgb} bind def
/col25 {0.630 0.250 0.000 srgb} bind def
/col26 {0.750 0.380 0.000 srgb} bind def
/col27 {1.000 0.500 0.500 srgb} bind def
/col28 {1.000 0.630 0.630 srgb} bind def
/col29 {1.000 0.750 0.750 srgb} bind def
/col30 {1.000 0.880 0.880 srgb} bind def
/col31 {1.000 0.840 0.000 srgb} bind def

end
save
-90.0 290.0 translate
1 -1 scale

/cp {closepath} bind def
/ef {eofill} bind def
/gr {grestore} bind def
/gs {gsave} bind def
/sa {save} bind def
/rs {restore} bind def
/l {lineto} bind def
/m {moveto} bind def
/rm {rmoveto} bind def
/n {newpath} bind def
/s {stroke} bind def
/sh {show} bind def
/slc {setlinecap} bind def
/slj {setlinejoin} bind def
/slw {setlinewidth} bind def
/srgb {setrgbcolor} bind def
/rot {rotate} bind def
/sc {scale} bind def
/sd {setdash} bind def
/ff {findfont} bind def
/sf {setfont} bind def
/scf {scalefont} bind def
/sw {stringwidth} bind def
/tr {translate} bind def
/tnt {dup dup currentrgbcolor
  4 -2 roll dup 1 exch sub 3 -1 roll mul add
  4 -2 roll dup 1 exch sub 3 -1 roll mul add
  4 -2 roll dup 1 exch sub 3 -1 roll mul add srgb}
  bind def
/shd {dup dup currentrgbcolor 4 -2 roll mul 4 -2 roll mul
  4 -2 roll mul srgb} bind def
/$F2psBegin {$F2psDict begin /$F2psEnteredState save def} def
/$F2psEnd {$F2psEnteredState restore end} def
%%EndProlog

$F2psBegin
10 setmiterlimit
n 0 842 m 0 0 l 595 0 l 595 842 l cp clip
 0.06000 0.06000 sc
7.500 slw
% Polyline
n 3075 1800 m 3600 1800 l gs col-1 s gr 
% Polyline
n 3075 1725 m 3600 1275 l gs col-1 s gr 
% Polyline
n 3075 1875 m 3600 2325 l gs col-1 s gr 
% Polyline
n 4200 1200 m 4650 1200 l gs col-1 s gr 
% Polyline
n 4200 1800 m 4650 1800 l gs col-1 s gr 
% Polyline
n 4200 2400 m 4650 2400 l gs col-1 s gr 
% Polyline
n 3075 4200 m 3600 4200 l gs col-1 s gr 
% Polyline
n 3059 4294 m 3584 4744 l gs col-1 s gr 
% Polyline
n 3059 4106 m 3584 3656 l gs col-1 s gr 
% Polyline
n 4211 3663 m 4736 4113 l gs col-1 s gr 
% Polyline
n 4200 4200 m 4725 4200 l gs col-1 s gr 
% Polyline
n 4179 4775 m 4704 4325 l gs col-1 s gr 
/Times-Roman ff 180.00 scf sf
1500 900 m
gs 1 -1 sc (1. ) col-1 sh gr
/Times-Roman ff 180.00 scf sf
2100 1875 m
gs 1 -1 sc (Principal \(x\) ) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 1800 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 2400 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 1275 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
4800 1200 m
gs 1 -1 sc (Presentity \(@PUA\)) col-1 sh gr
/Times-Roman ff 180.00 scf sf
4800 1800 m
gs 1 -1 sc (Presentity \(@PUA\)) col-1 sh gr
/Times-Roman ff 180.00 scf sf
4800 2400 m
gs 1 -1 sc (Presentity \(@PUA\)) col-1 sh gr
/Times-Roman ff 180.00 scf sf
1500 3300 m
gs 1 -1 sc (.2.) col-1 sh gr
/Times-Roman ff 180.00 scf sf
2100 4200 m
gs 1 -1 sc (Principal \(x\) ) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 3675 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 4275 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
3750 4800 m
gs 1 -1 sc (PUA) col-1 sh gr
/Times-Roman ff 180.00 scf sf
4875 4200 m
gs 1 -1 sc (Presentity \(@ ?? \)) col-1 sh gr
$F2psEnd
rs

--------------EA092FC0C53693519E1A26AE--

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