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Date: Thu, 30 Sep 2004 21:08:47 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Christian Stredicke <Christian.Stredicke@snom.de>
Subject: Re: [Sip] Symmetric NAT issue
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Christian Stredicke wrote:

> IMVHO SBC that use ICE are a smart alternative to using TURN. 

I don't really understand what this means. I would not expect an SBC to 
implement ICE, in the sense that it would do the p2p STUN pings, 
connectivity checks, etc., or are you proposing that it does?

Also, ICE makes use of TURN, to deal with the symmetric NAT case. SO I 
don't understand what you mean when you say ICE is an alternative to 
TURN. Can you explain?


UA that
> are not aware of ICE, STUN, TURN or NAT will run media through the B2BUA
> while ICE-aware devices can bypass them.

That I agree with.

> IMEHO ist much easier to
> implement STUN/ICE than TURN (well, we did/tried it) and it solves the
> problem with the "dumb" devices.

ICE without TURN will not help in the symmetric NAT case.


> Paired with DNS SRV and STUN its
> possible to automatically detect the nearest SBC, in case the SBC is
> necessary to relay media for symmetrical NAT.

Detect an SBC? I don't follow.

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: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Proposed bug fix to RFC3261 arising from draft-elwell-s
	ip-state-u pdate-02
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Paul,

Thanks for your comment. You have made it clear before that there is a
distinction between call state and dialog state, and that there is not
always a 1:1 relationship between them.

However, there are moves to deprecate dialog re-use. So although there are
examples of dialog re-use today (the REFER case being the prime example), I
don't anticipate new applications of this technique to be endorsed and there
is likely to be a move to phase out existing forms of re-use. 

Moreover there cannot be more than one call per dialog. Therefore if a
dialog supports a call, there is not a lot of difference in principle
between dialog state and call state, although if the dialog also supports a
subscription, call state information will not survive after a BYE, even
though dialog state will survive if the subscription survives.

With these considerations in mind, perhaps we don't need to put an excessive
amount of effort into distinguishing between call state and dialog state.
Call state just becomes a part of dialog state, existing on dialogs that
support calls but not on dialogs that support subscriptions.

I eliminated some of the elements such as Call-Info, the updating of which
did not appear to be of great interest to anyone and which, as you say, are
call state. Yes, session information I guess is also call state.

I forgot to add Required/Supported/Allowed headers, which I guess are in
fact part of the dialog state. From other specifications, PAI is an example
of data that is part of dialog state.

I guess the main question we need to answer is whether this proposed
clarification to RFC3261 (and eventually RFC3311) should embrace dialog
state change, call state change or both. One difficulty is that the term
"call" is not well-defined in RFC3261 (the Call-Id header unfortunately is
dialog state, not call state). Then the question arises whether we should
embark on trying to define the term call and hence call state.

So I see the following options, and I would like some feedback from the
list:

1. Do nothing.

2. Do not attempt to distinguish between call state and dialog state, on the
basis that the desire is to phase out dialog re-use (some dialogs will of
course also be calls, but a dialog should not serve two purposes:
subscription and call). In other words, continue as proposed in my previous
email (possibly restoring things like Call-Info if there is interest).

3. Stick rigidly to dialog state (so session information would be removed
from my proposal).

4. Define "call" and "call state" and distinguish them from "dialog" and
"dialog state". Handle both call state update and dialog state update.

Opinions please.

John Elwell (john.elwell@siemens.com)

Paul Kyzivat wrote:
I agree with what you are trying to do. But one rather significant 
detail with what you propose is that I don't believe the session 
description (SDP) is dialog state. It is call state. (A dialog that 
exists for the purpose of a SUBSCRIBE would not have this.)

That in some sense guts this proposal, since now there are no examples 
in this document of additional dialog state. However I still think the 
clarification is needed.

In addition, 3261 sweeps the distinction between dialog state and call 
state under the rug. Once alternative uses of dialogs are in play it is 
important to make this distinction clear, and to identify those items 
that are part of call state. This includes the session description info 
(the negotiated offer and answer as well as potentially another proposed 
change to offer and answer). It also includes the data associated with 
session timer, and possibly Call-Info, Alert-Info, Subject. (Which if 
any of these is truly call state remains to be discussed.)

	Paul

Elwell, John wrote:
> At the SIP meeting in San Diego, I was asked to raise on the mailing list
> the individual bug fixes proposed in the state-update draft, so that each
> can be given due scrutiny. I decided to start with RFC3261. Here only a
few
> changes are proposed and it does not seem sensible to consider these
> separately, so I propose them here as a single bug fix. If we can gain
> consensus on those, I will follow it up with messages proposing bug fixes
to
> the other documents (mainly RFC3311).
> 
> As a result of comments by Paul Kyzivat raised on the list recently, I
have
> modified the proposed changes slightly. In particular, I have dropped any
> headers that are not obviously dialog state changes, including those that
> might not be considered state at all (e.g., Alert-Info) and those that
might
> be considered call state rather than dialog state (e.g., Call-Info,
> Subject). If anyone wants these re-instating, please say so.
> 
> Proposed bug fix to RFC3261:
> 
> Add the following paragraph to the end of 12 (prior to 12.1):
> 
> "A dialog can also have certain other state information that can change
> during the lifetime of the dialog. This includes the session description
(as
> negotiated by SDP offer/answer), information from certain other headers
and
> information from certain other bodies. There are no examples from this
RFC.
> However, extensions may define other headers or bodies containing
> information that contributes to the state of a dialog and can change
during
> the lifetime of the dialog."
> 
> Add two new paragraphs to the end of 12.2 (prior to 12.2.1):
> 
> "A request, whether or not it is a target refresh request, MAY update
> certain other information transmitted at the time of dialog establishment.
> Although there are no examples in this RFC, other extensions may define
> headers or bodies that are appropriate for updating during a dialog.
> 
> "In this RFC, the only request defined that is suitable for updating
> information during a dialog is re-INVITE (see section 14). Other
extensions
> may define different requests suitable for updating information during a
> dialog."
> 
> Add to the end of the first paragraph of 14.1:
> 
> "A UAC MAY send a session description that is unchanged, if the purpose of
> the re-INVITE is solely for updating dialog-related information."
> 
> John Elwell (john.elwell@siemens.com)
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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	Mon, 27 Sep 2004 19:48:26 
Message-ID: <41582130.2060000@mistralsoftware.com>
Date: Mon, 27 Sep 2004 19:48:24 +0530
From: Gautham <gautham@mistralsoftware.com>
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To: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
Subject: Re: [Sip] Problem with an intial SIP INVITE server transa	ction.
References: <53F74F5A7B94D511841C00B0D0AB16F803E52FE9@baker.datcon.co.uk>
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Hi,

The scenario given by Roman could happen at either a proxy or a endpoint.

At a proxy, the appropriate thing would be to create a new transaction.
Like Roman has mentioned in his mail, this retransmission may be because 
the
1xx responses and the 2xx responses were lost. In such a case,  I believe,
it would be correct for the proxy to create a new transaction and
forward the INVITE downstream, as if it were a new transaction.
In anycase,  a proxy would not know better.

At an endpoint too, it should not be a problem if a new server 
transaction gets created
for the retransmitted INVITE. The UA core can look at the From  tag, 
Call-Id and possibly CSeq
to determine that it is not a new dialog request i.e., that it matches 
an ongoing dialog.
If the UA core is still retransmitting the 2xx, it would send the 2xx to 
the
transaction which would terminate it. If the UA core has finished 
retransmitting
the 2xx (and has received the ACK) it could treat the INVITE as an
out-of-sequence request.

Hope this helps.

Regards,
A. N. Gautham

Paul D.Smith wrote:

>Roman,
>
>I believe you have strayed into a poorly specified area of SIP here.  I have
>also found nothing in the specs. that directly answers your question however
>I can tell you what is observed:  the duplication INVITEs are simply
>ignored.  One option is to keep track of the INVITE transaction until the
>ACK arrives.  Although logically there is no transaction layer interaction
>(as per section 17, RFC 3261), you can then identify the INVITE as a
>duplicate and discard it.
>
>Paul D Smith
>Network Protocols Group 
>Data Connection Ltd (DCL)
>Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
>Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
>
>
>-----Original Message-----
>From: Roman Shpount [mailto:roman@atelo.com]
>Sent: 19 September 2004 08:15
>To: SIP Public Folder (E-mail)
>Subject: Re: RE: RE: [Sip] Problem with an intial SIP INVITE server
>transaction.
>
>
>Thomas,
>
>The problem is that retransmitted INVITE message does not have the To-tag.
>Thus it does not match the dialog and UAC is not even informed about this
>message. The only way I could think to deal with this, as I described in my
>original email, was to add a timer which keeps the server INVITE transaction
>open for T4 after 2XX response is sent. If retransmitted INVITE message
>arrives within this time, it matches the transaction and is ignored. By the
>way, this discussion is not academic, this is the real problem that we
>encountered while dealing with large volumes of SIP transactions over Wi-Fi.
>___________________________________
>Roman Shpount, VP of Technology
>aTelo, Inc. -- www.atelo.com
>
>------ Original message ------
>
>  
>
>>From: Thomas Gal <ThomasGal@LumenVox.com>
>>Subject: RE: RE: [Sip] Problem with an intial SIP INVITE server
>>    
>>
>transaction.
>  
>
>>Date: 31 Aug 04, 04:44 PM
>>To: 'Roman Shpount' <roman@atelo.com>,  <gledgard@iperia.com>,
>>    
>>
><sip@ietf.org>
>
>Maybe some people were just trying to help without scouring the spec for all
>the details. I have read the spec quite a few times, thanks, but here, maybe
>this will help(17):
>
>	In the case of a transaction where the
>   request was an INVITE (known as an INVITE transaction), the
>   transaction also includes the ACK only if the final response was not
>   a 2xx response.  If the response was a 2xx, the ACK is not considered
>   part of the transaction.
>
>      The reason for this separation is rooted in the importance of
>      delivering all 200 (OK) responses to an INVITE to the UAC.  To
>      deliver them all to the UAC, the UAS alone takes responsibility
>      for retransmitting them (see Section 13.3.1.4), and the UAC alone
>      takes responsibility for acknowledging them with ACK (see Section
>      13.2.2.4).  Since this ACK is retransmitted only by the UAC, it is
>      effectively considered its own transaction.
>
>You gotta wait til you get the ACK.
>
>-Tom
>
>
>-----Original Message-----
>From: Roman Shpount [mailto:roman@atelo.com] 
>Sent: Tuesday, August 31, 2004 8:38 AM
>To: gledgard@iperia.com; ThomasGal@LumenVox.com; sip@ietf.org
>Subject: Re: RE: [Sip] Problem with an intial SIP INVITE server transaction.
>
>Hi All, 
>
>I got quite a few answers like: "there is a magic key which prevents this"
>or "there is a timer that prevents this", or "you cannot get an INVITE once
>you sent the 1XX or 2XX response". I would suggest that you would read the
>RFC (Section 17.2.1) before rushing to answer my question. RFC clearly
>spells out that INVITE server transaction is terminated when 2XX response is
>sent. There is even a picture (Figure 7) that shows just that. There are no
>timers. There are no special transaction IDs. Transaction is ended, clear
>and simple. Once transaction is ended, re-transmitted INVITE message should
>start the new transaction.  
>___________________________________
>Roman Shpount, VP of Technology
>aTelo, Inc. -- www.atelo.com
>
>------ Original message ------
>
>  
>
>>From: Gordon Ledgard <gledgard@iperia.com>
>>Subject: RE: [Sip] Problem with an intial SIP INVITE server transaction.
>>Date: 31 Aug 04, 12:40 PM
>>To:  <ThomasGal@LumenVox.com>, Roman Shpount <roman@atelo.com>,  
>><sip@ietf.org>
>>    
>>
>
>
>Read the stuff about timers in the transaction layer. It spells out how you
>need to hold on to transactions for messages over unreliable transports for
>just this sort of reason.
>
>
>Gordon
>
>
>
>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of Thomas
>Gal
>Sent: Monday, August 30, 2004 7:54 PM
>To: 'Roman Shpount'; sip@ietf.org
>Subject: RE: [Sip] Problem with an intial SIP INVITE server transaction.
>
>
>I'm not an expert on this protocol (only read through the spec a few times
>so it blurs together with some of the other ones) but even without looking
>I'm SURE there's supposed to be a unique request-ID generated by the client
>to avoid just such a confusion. Of course in this case any stateful UA would
>ignore the already processed invite request. Also (sure about this one) if
>the client received the provisional(100) response it wouldn't re-transmit
>the INVITE either as that's the whole point.
>
>-Tom
>
>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Roman
>Shpount
>Sent: Tuesday, August 17, 2004 7:05 PM
>To: sip@ietf.org
>Subject: [Sip] Problem with an intial SIP INVITE server transaction.
>
>I run into the problem with handling of retransmitted SIP INVITE message
>based on the RFC 3621. 
>
>Imagine the following situation
>
>INVITE 1
>----------------------->
>100
><-----------------------
>2XX
><-----------------------
>re-transmitted INVITE 1
>----------------------->
>
>Based on RFC 3261, server INVITE transaction terminates as soon as 2XX
>response is sent. This means that re-transmitted INVITE will be treated as a
>new transaction. This INVITE message will not match the existing dialog,
>since its To tag is empty. This means it will be treated as new dialog
>creating message and phone will treat this message as a new call, which is
>clearly not intended.
>
>This situation can occur if both 100 and 2XX responses were lost by the
>unreliable transport, and INVITE continues to be re-transmitted. The fact
>that, 100 and 2XX responses were lost is unknown to the server transaction
>and INVITE re-transmit can arrive before the 2XX re-transmit is sent. 
>
>Majority of the transaction state machines in SIP are designed to handle
>messages, that are received after the response is sent. It seems that
>something simular is required in this case as well. 
>
>In our implementation of SIP stack, server transaction is not terminated
>when the 2XX response is sent. Instead, transaction is moved into the
>"ConfirmedBis" state and timer, which is set to fire in T4, is started. If
>the stack receives an INVITE message which matches the server transaction in
>this state, it ignores it. If stack receives a 2XX response, which matches
>the server transaction, it passes it to TU. Once the T4 timer is fired, the
>transaction is terminated.
>
>There is also an additional issue related to this problem. It looks like,
>for the sake of consistency, some type of response should be sent
>immediately to the transport layer when INVITE re-transmit is received. If
>multiple 2XX responses has been sent, it is unclear what the behavior should
>be: should all the 2XX messages be sent, should the last 2XX message be
>resent or should the INVITE be plainly ignored. 
>
>___________________________________
>Roman Shpount, VP of Technology
>aTelo, Inc. -- www.atelo.com
>
>
>
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use
>sip-implementors@cs.columbia.edu for questions on current sip Use
>sipping@ietf.org for new developments on the application of sip
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use
>sip-implementors@cs.columbia.edu for questions on current sip Use
>sipping@ietf.org for new developments on the application of sip
>
>
>
>
>
>
>
>
>
>
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>
>  
>



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From sip-bounces@ietf.org  Fri Oct  1 13:42:23 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Christer Holmberg \(JO/LMF\)'" <christer.holmberg@ericsson.com>,
        "'IETF SIP Mailing List'" <sip@ietf.org>
Subject: RE: [Sip] Question on RFC3261 Merged Requests
Date: Fri, 1 Oct 2004 12:02:26 -0400
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Christer,

I appreciate your comments. After thinking over the point you raised =
(why
old transaction mapping rules have to be used in 3261-compliant
environment), I came to the conclusion that 8.2.2.2 is completely =
broken.=20

My gateway example showed that at least when R-URI is different, second
INVITE should not be automatically rejected on the basis that it shares =
same
call-id, cseq, and the from tag with an INVITE in a different =
transaction.

Even when the two INVITEs have identical R-URI, the fact that they may =
have
visited different proxies after being forked could make one of them more
desirable than the other. The order they arrive at the UAS should not
determine which one gets presented to user and which one gets discarded. =
The
protocol can not make that distinction.

The forking proxy and calling UA both have built-in mechanisms to avoid =
one
single INVITE leading to two established dialogs. When it works, Section
8.2.2.2 is an optimization on top of that. And there are plenty of cases =
it
simply does not work.

I propose two options moving forward:

1. strike out 8.2.2.2 completely, or
2. replace it with a note to say that UAS may receive INVITEs in =
different
transactions but with identical call-id, cseq, from tag, and possibly =
same
R-URI. UAS should be prepared to handle this situation as it would with =
any
other incoming INVITEs.

I like to hear the list's opinion on this and whether it is a bug that
should be logged.

Regards,

Shan Lu
sentitO Networks =20

>-----Original Message-----
>From: Christer Holmberg (JO/LMF)=20
>[mailto:christer.holmberg@ericsson.com]=20
>Sent: Thursday, September 30, 2004 1:41 AM
>To: 'Shan Lu'; 'IETF SIP Mailing List'
>Subject: RE: [Sip] Question on RFC3261 Merged Requests
>
>
>
>Hi,
>
>Interesting point.
>
>One could argue that the existing text is ok for a normal SIP=20
>phone, because there is really no idea in receiving multiple=20
>"calls" (multiple "instances" of the same forked INVITE) from=20
>the same caller (in most cases the SIP phone can handle only=20
>one call at a time anyway, no matter if the incoming INVITEs=20
>origin from the same user or different users).
>
>However, for a gateway that may not be a valid case. Even if=20
>the gateway from a SIP perspective is a single SIP endpoint=20
>there are multiple end-users behind it, so IF we want to allow=20
>"forking" into the PSTN network (or whatever is on the other=20
>side) you are correct, I think.
>
>The problem I have with merged requests is that it takes away=20
>the advantages of only using the RFC3261 Via branch cookie=20
>from transaction identification. A node compliant with RFC3261=20
>will basically still have to use the "old" transaction mapping=20
>mechanism, in order the to the merge check...
>
>Regards,
>
>Christer Holmberg
>Ericsson Finland
>
>
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>> Shan Lu
>> Sent: 30. syyskuuta 2004 1:22
>> To: 'IETF SIP Mailing List'
>> Subject: [Sip] Question on RFC3261 Merged Requests
>>=20
>>=20
>> Hi,
>>=20
>> RFC3261 has following text:
>>=20
>> 8.2.2.2 Merged Requests
>>=20
>>    If the request has no tag in the To header field, the UAS=20
>core MUST
>>    check the request against ongoing transactions.  If the From tag,
>>    Call-ID, and CSeq exactly match those associated with an ongoing
>>    transaction, but the request does not match that=20
>transaction (based
>>    on the matching rules in Section 17.2.3), the UAS core SHOULD
>>    generate a 482 (Loop Detected) response and pass it to the server
>>    transaction.
>>=20
>> Assume that calls to support@example.com is forked to two=20
>> phone numbers on
>> PSTN, say, 555-1111 and 555-2222. Further assume that there=20
>> is only one
>> SIP-PSTN gateway. The two INVITEs arriving at the gw are=20
>> identical except
>> for R-URI userinfo and top VIA branch (assuming forking proxy=20
>> is the only
>> proxy).=20
>>=20
>> My question is: why should GW 482 second INVITE? Isn't that=20
>> the purpose of
>> forking? Should 8.2.2.2 be modified as:
>>=20
>> ... If request URI, the From tag, Call-ID, and CSeq exactly match ...
>>        ^^^^^^^^^^^^=20
>>=20
>> Regards,=20
>>=20
>> Shan Lu
>> sentitO Networks
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>=20
>


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Hello Everyone,

My last day at Cisco is October 8th.  I will be starting a new position 
at Airespace on November 1st. In the interim, you can contact me at the 
following email address:

	r o h a n ( a t ) e k a b a l ( d o t ) c o m

Please also be aware that I will be traveling in Guatemala from October 
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See you all in Washington, DC.

many thanks,
-rohan


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<fontfamily><param>Helvetica</param>Hello Everyone,


My last day at Cisco is October 8th.  I will be starting a new
position at Airespace on November 1st. In the interim, you can contact
me at the following email address:


	r o h a n ( a t ) e k a b a l ( d o t ) c o m


Please also be aware that I will be traveling in Guatemala from
October 12 till October 27 (Note that this period includes both draft
deadlines).  I will be only checking email irregularly during this
time.


See you all in Washington, DC.


many thanks,

-rohan

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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Subject: [Sip] History-Info: Processing of Reason header in responses 
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Hi all,

Per my posting on the SIPPING list, the currently agreed option for the
Reason header proposal for redirection would impact the History-Info
processing because it was never considered to capture the Reason header from
a response as part of the History-Info entries,  but rather the SIP response
codes are used to generate a new Reason header associated with a specific
retargeted-uri.  Independent of whether the WG progresses the redirection
reason proposal, there is a general issue as to whether the History-Info
should use the Reason header in the responses in the case of retargeting.
Per RFC 3328, you can have multiple Reason headers, however, only one SIP
reason header.  And, while the typical scenario for Reason header was for
requests, it doesn't preclude the use of the header in responses.  

So, it does seem in terms of completeness, History-Info should specify the
general processing if there is a Reason header in the response.  I'm
proposing to add some discussion to section 4.3.3.1.2 replacing the existing
1st sentence of the 1st paragraph:
"For retargets that are the result of an explicit SIP response, the SIP
Response Code that triggered the retargeting MUST be included in the Reason
header field of the Request URI that has been retargeted."

 with something like the following:
"For retargets that are the result of an explicit SIP response, a Reason
header field MUST be included.  If the SIP response does not include a
Reason header, the SIP Response Code that triggered the retargeting MUST be
included in the Reason header field of the Request URI that has been
retargeted.  If the response contains a non-SIP Reason header, it MUST be
captured as an additional Reason associated with the Request URI that has
been retargeted, along with the SIP Response Code.  If the Reason header is
a SIP reason, then it MUST be used rather than the SIP response code." 

I'd appreciate any concerns be raised by the end of this week (Oct. 8). 

Thanks,
Mary H. Barnes
mary.barnes@nortelnetworks.com


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

<P><FONT SIZE=3D2 FACE=3D"Arial">Per my posting on the SIPPING list, =
the currently agreed option for the Reason header proposal for =
redirection would impact the History-Info processing because it was =
never considered to capture the Reason header from a response as part =
of the History-Info entries,&nbsp; but rather the SIP response codes =
are used to generate a new Reason header associated with a specific =
retargeted-uri.&nbsp; Independent of whether the WG progresses the =
redirection reason proposal, there is a general issue as to whether the =
History-Info should use the Reason header in the responses in the case =
of retargeting.&nbsp; Per RFC 3328, you can have multiple Reason =
headers, however, only one SIP reason header.&nbsp; And, while the =
typical scenario for Reason header was for requests, it doesn't =
preclude the use of the header in responses.&nbsp; </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">So, it does seem in terms of =
completeness, History-Info should specify the general processing if =
there is a Reason header in the response.&nbsp; I'm proposing to add =
some discussion to section 4.3.3.1.2 replacing the existing 1st =
sentence of the 1st paragraph:</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&quot;For retargets that are the =
result of an explicit SIP response, the SIP Response Code that =
triggered the retargeting MUST be included in the Reason header field =
of the Request URI that has been retargeted.&quot;</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&nbsp;with something like the =
following:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&quot;For retargets that are the =
result of an explicit SIP response, a Reason header field MUST be =
included.&nbsp; If the SIP response does not include a Reason header, =
the SIP Response Code that triggered the retargeting MUST be included =
in the Reason header field of the Request URI that has been =
retargeted.&nbsp;</FONT> <FONT SIZE=3D2 FACE=3D"Arial">If the response =
contains a non-SIP Reason header, it MUST be captured as an additional =
Reason associated with the Request URI that has been retargeted, along =
with the SIP Response Code.&nbsp; If the Reason header is a SIP reason, =
then it MUST be used rather than the SIP response code.&quot; =
</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I'd appreciate any concerns be raised =
by the end of this week (Oct. 8). </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Mary H. Barnes</FONT>
<BR><I><FONT SIZE=3D1 =
FACE=3D"Arial">mary.barnes@nortelnetworks.com</FONT></I>
</P>

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From sip-bounces@ietf.org  Tue Oct  5 12:57:06 2004
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Subject: [Sip] SIP state diagram
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Is anyone aware of an existing state diagram for the current rfc?

thanks
moe smadi

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From sip-bounces@ietf.org  Tue Oct  5 14:37:59 2004
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Subject: [Sip] Anybody planning on submitting a new -00 WG draft soon?
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I've been asked to pre-approve any new working group 
(draft-ietf-sip-stuff-00) drafts in this submission cycle. Currently, 
I'm not aware of any. Anybody know better?

--
Dean


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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Subject: RE: FW: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915: trying 
	next	 transport upon failure
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Jonathan,

Thanks for the clarification regarding the use of NAPTR records.

Just so that we are 100% clear on the other cases, can you first confirm
that
an implementation is allowed to pick any transport it prefers, look for an 
SRV record and if one is found, use that transport?  And as per the NAPTR
example, if that transport fails, then no further transports are tried?

But if no SRV records were found for the chosen transport, another transport
may be chosen until either SRV records are found, or there are no further
transports to try.

And finally the final case of no SRV records.  In this case the transport to
use is UDP for SIP: URIs and TLS (over TCP) for SIPS: URIs.  Agreed?

Thanks,
Paul D Smith
Network Protocols Group 
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: 29 September 2004 07:38
To: SIP Public Folder (E-mail)
Subject: Re: FW: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915:
trying next transport upon failure


Thanks for prodding me to answer this.

I agree this is confusing. RFC 3261 seems to refer to the notion that 
you can change transports on different attempts (end of 8.1.2). However, 
RFC 3263 would seem to indicate that the NAPTR result would be a single 
service, and therefore, a single transport.

The correct behavior is the latter. That is, the NAPTR processing will 
select a single transport, and that's what gets used. Previously, we 
were allowing attempts across transports because we were combining 
addresses from different SRV records and merging them. This was deemed a 
no-no by the DNS folks, due to the operational issues with comparing 
preference values across records.

Once we eliminated this merging, the updated procedures in RFC 3263 
effectively resulted in selection of a single transport, but this wasn't 
  made fully consistent with RFC 3261. I've logged a bug against 3261 to 
clarify (http://bugs.sipit.net/sipwg/show_bug.cgi?id=760).

Thanks,
Jonathan R.

Paul D.Smith wrote:

> Jonathan,
> 
> Can I please ask you to comment on the discussion Brett and I have been
> having
> via the WG regarding the use of NAPTR records?
> 
> The question is whether the use of multiple transports, indicated by
> multiple
> NAPTR records, SRV records or client support for multiple transports
should
> be attempted or not.
> 
> I can see an advantage to defining, say "TLS preferred over TCP over UDP"
> but
> this would only be useful under the "just try one transport" idea if the
> client only supported one of the less preferred options.
> 
> However the "try all in order" idea allows for, for example, a separate
TLS
> connection server to be preferred but for TCP to be acceptable (possibly
> with
> some restrictions imposed through other means) if the TLS server is
> unavailable.
> 
> So what is the intended processing?  Also in either case, would you like
to 
> raise a bugzilla report to ensure this is clarified in a future release of
> the
> specification?
> 
> Thanks,
> Paul D Smith
> Network Protocols Group 
> Data Connection Ltd (DCL)
> Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com 
> Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com
> 
> 
> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: 17 September 2004 18:31
> To: SIP Public Folder (E-mail)
> Subject: RE: [Sip-implementors] RE: [Sip] rfc3263 and rfc2915: trying
> next transport upon failure
> 
> 
> Hi Jonathan,
> 
> Do you agree with Paul's interpretation 
> concerning advancement to the next transport
> (next NAPTR record or set of SRV records)?
> 
> My interpretation was that advancing
> between transports has been removed.
> 
> There are likely good reasons for and
> against advancing between transports.
> I am just trying to understand what
> rfc3263, rfc2915, and rfc3261 indicate
> or should have indicated.
> 
> Thanks in advance,
> Brett Tate
> 
> 
> 
>>RFC3263, section 4.1, contains the following text
>>
>> If no NAPTR records are found, the client 
>> constructs SRV queries for those transport 
>> protocols it supports, and does a query for 
>> each.  Queries are done using the service 
>> identifier "_sip" for SIP URIs and "_sips" 
>> for SIPS URIs.  A particular transport is 
>> supported if the query is successful.  The 
>> client MAY use any transport protocol it 
>> desires which is supported by the server.
>>
>>This means that potentially the set of 
>>SRV records covers multiple protocols in 
>>the case that no NAPTR records are returned.
> 
> 
> I interpret the text as indicating that one
> transport should be selected instead of
> potentially combining the multiple sets of
> srv records.
> 
> 
>>It therefore seems only sensible to me 
>>that if multiple transports are indicated 
>>by NAPTR records, that they too should be 
>>tried, in the order indicated by the NAPTR 
>>records.
> 
> 
> Hopefully there would have been a statement 
> indicating when to advance to the next NAPTR
> record.  Or there could have been a statement
> indicating the combining of all the SRV records
> for each NAPTR record.
> 

-- 
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: Juha Heinanen <jh@tutpro.com>
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i read draft-ietf-sip-identity-03 and noticed that in order to use it,
the domains must still have private/public certificates that are "valid
back to a trusted CA".

my domain tutpro.com has not purchased such certificate and i would bet
that the same holds true for a VERY large percent of all internet
domains.  as long as this is true, the proposed identity verification
solution may work in theory but not is practice.

when thinking about this, i started to wonder if it would be possible
for the UAS to verify the identity of the UAC from the domain via a few
sip requests, i.e., by breaking this requirement:

      User agents that receive identity assurances must be able to
      validate these assurances without performing any network lookup.

by the way, i don't think that this requirement holds true in the
proposed solution either, since the uas must acquire the certificate of
the signing domain, which may mean network lookup.

so how about a certificate free system where the UAS authenticates the
request with the domain by sending to the proxy of the domain a
verification request?  when the proxy receives the verification request
it challenges the UAC and then replies to the UAS.  comments?

-- juha

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From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
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Hi,
 Can anybody provide some info on Far End Camera Control in SIP.

Regards
Rajeew


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Juha Heinanen wrote:

>
>so how about a certificate free system where the UAS authenticates the
>request with the domain by sending to the proxy of the domain a
>verification request?  when the proxy receives the verification request
>it challenges the UAC and then replies to the UAS.  comments?
>
>  
>
How would one authenticate the domain's proxy, such that a MITM could 
not substitute a proxy that would then falsely vouch for the UA?

The usual answer for this is "check the proxy's cert" which of course 
circles us back onto the original argument.

Other alternatives that might work would include something like a 
Kerberos model, but that leaves us with a problem of interdomain 
authentication which essentially boils down to the cert problem.

--
Dean

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From: Juha Heinanen <jh@tutpro.com>
To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] verifying identity of sender
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Dean Willis writes:

 > How would one authenticate the domain's proxy, such that a MITM could 
 > not substitute a proxy that would then falsely vouch for the UA?

yes, my proposal doesn't offer 100 % security. but considering how much
you get, it would still be good thing to do.  very few sip users today
authenticate the proxy they talk to.  if that would be have been a
requirement from the start, we would have very little sip usage in the
first place.

-- juha

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Juha Heinanen wrote:

>Dean Willis writes:
>
> > How would one authenticate the domain's proxy, such that a MITM could 
> > not substitute a proxy that would then falsely vouch for the UA?
>
>yes, my proposal doesn't offer 100 % security. but considering how much
>you get, it would still be good thing to do.  very few sip users today
>authenticate the proxy they talk to.  if that would be have been a
>requirement from the start, we would have very little sip usage in the
>first place.
>  
>
Why? It works for HTTP. You wouldn't think of buying from a web site, 
doing banking, stock trades, or anything else sensitive without looking 
for the little padlock icon, would you?  And if somebodody else did, and 
got ripped off, you'd laugh at them, right?. Well, I would, but you're 
probably a nicer guy and would just point out their security fault.

--
Dean



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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Juha Heinanen'" <jh@tutpro.com>, sip@ietf.org
Subject: RE: [Sip] verifying identity of sender
Date: Wed, 6 Oct 2004 14:27:04 -0400 
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Juha,

Fundamentally, in the only context in which certificates are widely used
today, a certificate is a contract that binds a set of keying material to a
domain name. What we need here for sip-identity is a way of binding a set of
keying material to a domain name. There are, of course, alternatives to
certificates, like putting keying material in the DNS, and, I guess, relying
on the DNS to "make a few SIP requests", as you suggest. However, the
insecurity of the DNS at the moment makes these sorts of approaches
problematic. Also, I think the security community is deeply suspicious of
attempts to reinvent the certificate construct in some other way. This is
what certificates are for: if we need this property, we should use
certificates.

Perhaps even more fundamentally, without TLS, sip-identity-03 would not work
at all. TLS assumes that your SIP proxies have a domain certificate. People
must be able to verify this certificate. TLS is mandatory-to-implement for
SIP proxy servers according to RFC3261. In other words, I'm not requiring
anything that isn't required of proxy servers by RFC3261 already. Finally, I
don't really think certificates are prohibitively expensive, and so on.

> by the way, i don't think that this requirement holds true in the
> proposed solution either, since the uas must acquire the certificate of
> the signing domain, which may mean network lookup.

Yes, it may mean network lookup, but doesn't necessarily. There are
certainly other ways for the UAS to acquire certificates. This system does
not require a network lookup to work because certificates are
self-verifying. Any system that relies on a network lookup is troublesome,
generally, because caching credentials because very problematic. That much
said, probably I should probably make the phrasing of that requirement more
precise.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Wednesday, October 06, 2004 1:30 AM
> To: sip@ietf.org
> Subject: [Sip] verifying identity of sender
> 
> 
> i read draft-ietf-sip-identity-03 and noticed that in order to use it,
> the domains must still have private/public certificates that are "valid
> back to a trusted CA".
> 
> my domain tutpro.com has not purchased such certificate and i would bet
> that the same holds true for a VERY large percent of all internet
> domains.  as long as this is true, the proposed identity verification
> solution may work in theory but not is practice.
> 
> when thinking about this, i started to wonder if it would be possible
> for the UAS to verify the identity of the UAC from the domain via a few
> sip requests, i.e., by breaking this requirement:
> 
>       User agents that receive identity assurances must be able to
>       validate these assurances without performing any network lookup.
> 
> by the way, i don't think that this requirement holds true in the
> proposed solution either, since the uas must acquire the certificate of
> the signing domain, which may mean network lookup.
> 
> so how about a certificate free system where the UAS authenticates the
> request with the domain by sending to the proxy of the domain a
> verification request?  when the proxy receives the verification request
> it challenges the UAC and then replies to the UAS.  comments?
> 
> -- juha
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: Juha Heinanen <jh@tutpro.com>
To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] verifying identity of sender
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Dean Willis writes:

 > Why? It works for HTTP. You wouldn't think of buying from a web site, 
 > doing banking, stock trades, or anything else sensitive without looking 
 > for the little padlock icon, would you?  And if somebodody else did, and 
 > got ripped off, you'd laugh at them, right?. Well, I would, but you're 
 > probably a nicer guy and would just point out their security fault.

online shopping is very much different application than sip
applications.  when you sell something, you can justify going through
the trouble of buying a certificate (although i have heard that those
are available under very loose basis and don't really prove much), but i
don't see it a viable option for each sip proxy.  take email as an
example.  it is an old application and still very few email servers use
certificates to verify senders.

i just repeat that a certificate free identity solution would prevent a
very large percent of sip requests coming from faked senders.  it would
not protect against mitm attacks or dns hijacking, but those are much
more difficult things to do that simply faking the from field of the
request that is within the reach of anybody.  it would thus make more
sense to me to have a imperfect deployable solution rather than a
"correct" solution that very few can deploy.

-- juha

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From: Juha Heinanen <jh@tutpro.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>
Subject: RE: [Sip] verifying identity of sender
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Peterson, Jon writes:

 > However, the
 > insecurity of the DNS at the moment makes these sorts of approaches
 > problematic. Also, I think the security community is deeply suspicious of
 > attempts to reinvent the certificate construct in some other way. This is
 > what certificates are for: if we need this property, we should use
 > certificates.

i didn't try to propose any new certificate construct.  uas would just
send a verification request to the proxy of the domain and would get
back a reply.

 > Perhaps even more fundamentally, without TLS, sip-identity-03 would not work
 > at all. TLS assumes that your SIP proxies have a domain certificate. People
 > must be able to verify this certificate. TLS is mandatory-to-implement for
 > SIP proxy servers according to RFC3261. In other words, I'm not requiring
 > anything that isn't required of proxy servers by RFC3261 already. Finally, I
 > don't really think certificates are prohibitively expensive, and so
 > on.

implementation of tls in the proxy is one thing and getting a
certificate for it that is verifiable to a known CA is another thing.  i
can imagine that large operators can do it, but not small companies and
private people who use sip more or less in peer-to-peer mode.

-- juha

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From: Dean Willis <dean.willis@softarmor.com>
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Peterson, Jon wrote:

>Yes, it may mean network lookup, but doesn't necessarily. There are
>certainly other ways for the UAS to acquire certificates. This system does
>not require a network lookup to work because certificates are
>self-verifying. Any system that relies on a network lookup is troublesome,
>generally, because caching credentials because very problematic. That much
>said, probably I should probably make the phrasing of that requirement more
>precise.
>  
>
Checking certificate revocation lists, which is an important part of 
TLS, does seem to require network lookup.

--
Dean


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Juha Heinanen wrote:

>i just repeat that a certificate free identity solution would prevent a
>very large percent of sip requests coming from faked senders.  it would
>not protect against mitm attacks or dns hijacking, but those are much
>more difficult things to do that simply faking the from field of the
>request that is within the reach of anybody.  it would thus make more
>sense to me to have a imperfect deployable solution rather than a
>"correct" solution that very few can deploy.
>
>  
>

It seems this has a lot in common with the MARID email work, which as I 
recall tried two approaches:

1) Putting keying material into DNS that a domain server could use to 
"sign" outbound email, thereby proving it had come from that domain
2) developing a sort of "outbound" MX record for DNS that indicates 
authoritative sender relays for a domain, allowing receivers to verify 
(by address matching) that the message had at least come through a 
server in the indicated domain.

Both approaches are transitive trust models, with DNS dependencies, that 
vary by the strength of the trust mechanism.

If one wished to "water down" identity for deployability one could 
essentially use the 3GPP approach of P-Asserted-Identity combined with 
veryifying that the Via header contains an entry listed in DNS as an 
authoritative proxy for the domain asserted in the P-Asserted-Identity 
header field, and trusting that said proxy enforced identity. This is a 
little lighter than polling said proxy, but does slightly less about 
MITMs, as another proxy in the chain might have changed the P-Asserted 
-Identity header field value.

It may also be interesting to note that MARID apparently gave up in 
disgust at the patent situation.

--
Dean

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From sip-bounces@ietf.org  Wed Oct  6 19:13:47 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Juha Heinanen'" <jh@tutpro.com>
Subject: RE: [Sip] verifying identity of sender
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Juha,

>  > However, the
>  > insecurity of the DNS at the moment makes these sorts of approaches
>  > problematic. Also, I think the security community is deeply suspicious
of
>  > attempts to reinvent the certificate construct in some other way. This
is
>  > what certificates are for: if we need this property, we should use
>  > certificates.
> 
> i didn't try to propose any new certificate construct.  uas would just
> send a verification request to the proxy of the domain and would get
> back a reply.

I didn't mean that you were proposing a new model for certificates - I meant
that you were trying to propose a new way of getting the same properties
that we get from certificates. I think that is unnecessary and potentially
very dangerously.

> implementation of tls in the proxy is one thing and getting a
> certificate for it that is verifiable to a known CA is another thing.  i
> can imagine that large operators can do it, but not small companies and
> private people who use sip more or less in peer-to-peer mode.
> 

Anyway, implementation has nothing to do with this. If you don't -do- TLS,
you can't provide an Identity header; or rather, the assertion that Identity
provides is valueless.

Look, this isn't something where every user of a domain needs their own
certificate, or something. This is a case where you need one cert for one
domain. How much do certs cost? Probably as much as $350 a year, down to as
little as $40 dollars a year. I'm sorry, but compared to the cost of
operating a server, bandwidth, etc etc, I don't think this is such a
tremendous impediment.

Jon Peterson
NeuStar, Inc.

> -- juha
> 

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Date: Thu, 7 Oct 2004 08:21:59 +0300
From: Juha Heinanen <jh@tutpro.com>
To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] verifying identity of sender
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Dean Willis writes:

 > 1) Putting keying material into DNS that a domain server could use to 
 > "sign" outbound email, thereby proving it had come from that domain
 > 2) developing a sort of "outbound" MX record for DNS that indicates 
 > authoritative sender relays for a domain, allowing receivers to verify 
 > (by address matching) that the message had at least come through a 
 > server in the indicated domain.

here nothing new needs to be put in dns.  when uas receives invite, it
gives back a provisional response, constructs the verify request
including a hash of invite's from uri, call id and from tag, and then
sends it out like any out-of-dialog request using the from uri as
request uri.  when the uac of the invite receives the verify request, it
checks if it has any pending invite that matches the hash and responds
either with 200 ok or some failure code.  when uas receives the verify
response it gives final response to the invite and that is it.

-- juha

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From sip-bounces@ietf.org  Thu Oct  7 09:01:54 2004
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From: vikram <vikramb@aftek.com>
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Subject: [Sip] Nat traversal !
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 Hello all ,
 
  For symmetric NAT traversal , I tried  my VoIP client with some free  
SBCs (termed as Out bound proxies also )
  I could register with them and even call is established but there was 
no voice behind Symmetric NAT.

  1. In the source code I am sending all my packets (sip as well as rtp) 
to SBCs's IP & port
     We are using same socket to send/recv rtp packet. It this enough ?
   2. Do I have to do anything  additionally  ?
  3. With the same set up X-lite(xten) worked , but our s/w could not .
  4. we have compared x-lite packets with ours,  they are very same.    

When we sniff  packets at our ISP we could see packets coming in
but our Symmetric NAT -router might be  dropping them .

If router  blocks some traffic because of its symmetric Nature
dose it respond with ICMP  error messages ? how do i know
that my router is dropping (specially when my router is a old and
low end )

This is happening even if we are using a SBC.
Our logic :
1.Check NAT type
2. If it is Symmetric NAT
        Send local IP & media  port in  SDP.
        Use SBC IP & port while sending  anything (sip/rtp)

any guess what is going wrong ?

Please HELP !!!!!

regards

vikram

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	Thu, 07 Oct 2004 07:25:43 PDT
Date: Thu, 7 Oct 2004 07:25:43 -0700 (PDT)
From: Ranjit Avasarala <ranjsadh@yahoo.com>
Subject: Re: [Sip] Far End Camera Control in SIP
To: rajeewsingh@tataelxsi.co.in
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Hi rajeev

 SIP is not the protocol of choice for far end camera
control or for video conferencing. More ideal protocol
 would be H.324 and H.324M for 3GPP. 

Ranjit

--- Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
wrote:

> 
> Hi,
>  Can anybody provide some info on Far End Camera
> Control in SIP.
> 
> Regards
> Rajeew
> 
> 
> _______________________________________________
> Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip
> 


=====
Regards
Ranjit


		
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Date: Thu, 07 Oct 2004 08:26:48 -0700
Subject: Re: [Sip] verifying identity of sender
From: Cullen Jennings <fluffy@cisco.com>
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If you want a free certificate, have a look at http://cacert.org

It sounds like the sort of scheme you discuss below would end up relying on
the security of DNS which, short of DNSSec, is not very good. Certificates
are not that hard to get but yes I agree they do take some effort. The
problem is, no one has proposed an alternative solutions that does not use
them and looks deployable.  The current identity scheme adds a little extra
work for the domain administrator and a close to zero cost. It does not add
any additional per end user cost or effort. This seems pretty good compared
to any alternatives that have been proposed.

Cullen



On 10/6/04 1:30 AM, "Juha Heinanen" <jh@tutpro.com> wrote:

> i read draft-ietf-sip-identity-03 and noticed that in order to use it,
> the domains must still have private/public certificates that are "valid
> back to a trusted CA".
> 
> my domain tutpro.com has not purchased such certificate and i would bet
> that the same holds true for a VERY large percent of all internet
> domains.  as long as this is true, the proposed identity verification
> solution may work in theory but not is practice.
> 
> when thinking about this, i started to wonder if it would be possible
> for the UAS to verify the identity of the UAC from the domain via a few
> sip requests, i.e., by breaking this requirement:
> 
>       User agents that receive identity assurances must be able to
>       validate these assurances without performing any network lookup.
> 
> by the way, i don't think that this requirement holds true in the
> proposed solution either, since the uas must acquire the certificate of
> the signing domain, which may mean network lookup.
> 
> so how about a certificate free system where the UAS authenticates the
> request with the domain by sending to the proxy of the domain a
> verification request?  when the proxy receives the verification request
> it challenges the UAC and then replies to the UAS.  comments?
> 
> -- juha
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Date: Thu, 07 Oct 2004 08:58:37 -0700
Subject: Re: [Sip] verifying identity of sender
From: Cullen Jennings <fluffy@cisco.com>
To: Juha Heinanen <jh@tutpro.com>, Dean Willis <dean.willis@softarmor.com>
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On 10/6/04 10:21 PM, "Juha Heinanen" <jh@tutpro.com> wrote:

> Dean Willis writes:
> 
>> 1) Putting keying material into DNS that a domain server could use to
>> "sign" outbound email, thereby proving it had come from that domain
>> 2) developing a sort of "outbound" MX record for DNS that indicates
>> authoritative sender relays for a domain, allowing receivers to verify
>> (by address matching) that the message had at least come through a
>> server in the indicated domain.
> 
> here nothing new needs to be put in dns.  when uas receives invite, it
> gives back a provisional response, constructs the verify request
> including a hash of invite's from uri, call id and from tag, and then
> sends it out like any out-of-dialog request using the from uri as
> request uri.  when the uac of the invite receives the verify request, it
> checks if it has any pending invite that matches the hash and responds
> either with 200 ok or some failure code.  when uas receives the verify
> response it gives final response to the invite and that is it.
> 

ok so my from says if X send you a message, then X can receive your reply.
How does this help know who X is?




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From sip-bounces@ietf.org  Thu Oct  7 17:55:05 2004
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Cullen Jennings wrote:

>>here nothing new needs to be put in dns.  when uas receives invite, it
>>gives back a provisional response, constructs the verify request
>>including a hash of invite's from uri, call id and from tag, and then
>>sends it out like any out-of-dialog request using the from uri as
>>request uri.  when the uac of the invite receives the verify request, it
>>checks if it has any pending invite that matches the hash and responds
>>either with 200 ok or some failure code.  when uas receives the verify
>>response it gives final response to the invite and that is it.
>>
>>    
>>
>
>ok so my from says if X send you a message, then X can receive your reply.
>How does this help know who X is?
>  
>
Not quite.

I INVITE you to a call.  Upon sending the INVITE, my UA sends a PUBLISH 
to the softarmor.com server saying I really did send an INVITE to you, 
and including a hash of that INVITE. Hopefully, softarmor.com 
authenticates that PUBLISH message.

You get the invite, wonder if  it really came from someone who could 
authenticate as me to softarmor.com, poll that server (including the 
hash) and softarmor.com tells you "Yes, that INVITE came from somebody 
who could authenticate as Dean. Trust me."

So, the strength of the assertion of softarmor.com as to my identity is 
contingent on two things:

1) The strength of your belief that the server you think is 
softarmor.com really is (and isn't somebody else just faking)
2) The strength of your belief in the functional integrity and 
authentication provided by softarmor.com

In the simplest case, all that this technique proves is that whoever 
sent you the message is either 1)  authenticated to the asserting 
server, or 2) capable of munging DNS or IP routing to intercept your 
request to the asserting server, making them sufficiently interesting to 
talk to anyhow.

In the typical "dumb ISP" case, this is a weak assertion, but it's still 
much stronger than we get for email and would block the majority of 
source-fraud spam.

But in a confederated environment, where servers might have IPSEC SAs or 
be linked by private, protected network, the assertion could be a bit 
stronger.


Questions:

1) Is this imperfect solution so useful as to be worth specifying, even 
though we know it's not complete?

2) Is the benefit of the solution worth the additional 4 message 
overhead on every transaction?

3) Are there obvious  optimizations, such as having the asserting server 
"learn" about the request by observation rather than explicit notification?

4) Could we just use a MARID approach, and configure "edge" proxies to 
discard inbound requests where the source IP address doesn't show in DNS 
as authoritative for the domain component of the From: address, thereby 
getting most of the benefit with less overhead?

--
Dean



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Date: Fri, 08 Oct 2004 11:29:35 +0530
From: Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
Subject: RE: [Sip] Far End Camera Control in SIP
In-reply-to: <20041007142543.23384.qmail@web53605.mail.yahoo.com>
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Hi Ranjit,
   Now people have started working on SIP based video conferencing.
What till now I came to know that the
"support for Far End Camera Control by SIP User Agent"
    can be signalled by SDP using the "m" field giving support for H.224
like

m:data  9000 RTP/AVP 100
Media Attribute(a):rtpmap:100 H224

This shows that the User Agent has support for Far End Camera Control.
Now the Far End Camera Control messages can be carried in RTP during the
conferencing.
The User agent should be intelligent enough to know the RTP streams related
to
Far End Camera Control..and accordingly the Camera is controlled.

Let me know if there is any misunderstanding on this part..

Thanks
Rajeew


-----Original Message-----
From: Ranjit Avasarala [mailto:ranjsadh@yahoo.com]
Sent: Thursday, October 07, 2004 7:56 PM
To: rajeewsingh@tataelxsi.co.in
Cc: sip@ietf.org
Subject: Re: [Sip] Far End Camera Control in SIP


Hi rajeev

 SIP is not the protocol of choice for far end camera
control or for video conferencing. More ideal protocol
 would be H.324 and H.324M for 3GPP.

Ranjit

--- Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
wrote:

>
> Hi,
>  Can anybody provide some info on Far End Camera
> Control in SIP.
>
> Regards
> Rajeew
>
>
> _______________________________________________
> Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip
>


=====
Regards
Ranjit



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Hi Rajeev

what u say is ok. I mean u can indicate support for 224 in SDP. But IP 
network is not capable of carrying delay sensitive information.

Ranjit



Rajeew Kumar Singh wrote:
> Hi Ranjit,
>    Now people have started working on SIP based video conferencing.
> What till now I came to know that the
> "support for Far End Camera Control by SIP User Agent"
>     can be signalled by SDP using the "m" field giving support for H.224
> like
> 
> m:data  9000 RTP/AVP 100
> Media Attribute(a):rtpmap:100 H224
> 
> This shows that the User Agent has support for Far End Camera Control.
> Now the Far End Camera Control messages can be carried in RTP during the
> conferencing.
> The User agent should be intelligent enough to know the RTP streams related
> to
> Far End Camera Control..and accordingly the Camera is controlled.
> 
> Let me know if there is any misunderstanding on this part..
> 
> Thanks
> Rajeew
> 
> 
> -----Original Message-----
> From: Ranjit Avasarala [mailto:ranjsadh@yahoo.com]
> Sent: Thursday, October 07, 2004 7:56 PM
> To: rajeewsingh@tataelxsi.co.in
> Cc: sip@ietf.org
> Subject: Re: [Sip] Far End Camera Control in SIP
> 
> 
> Hi rajeev
> 
>  SIP is not the protocol of choice for far end camera
> control or for video conferencing. More ideal protocol
>  would be H.324 and H.324M for 3GPP.
> 
> Ranjit
> 
> --- Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
> wrote:
> 
> 
>>Hi,
>> Can anybody provide some info on Far End Camera
>>Control in SIP.
>>
>>Regards
>>Rajeew
>>
>>
>>_______________________________________________
>>Sip mailing list
>>https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP
>>Protocol
>>Use sip-implementors@cs.columbia.edu for questions
>>on current sip
>>Use sipping@ietf.org for new developments on the
>>application of sip
>>
> 
> 
> 
> =====
> Regards
> Ranjit
> 
> 
> 
> _______________________________
> Do you Yahoo!?
> Declare Yourself - Register online to vote today!
> http://vote.yahoo.com
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


-- 
Regards
Ranjit





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From sip-bounces@ietf.org  Fri Oct  8 07:53:43 2004
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From: "Spencer Dawkins" <spencer@mcsr-labs.org>
To: <sip@ietf.org>
References: <BD8AB5BD.143BE%fluffy@cisco.com> <4165B3CF.8020707@softarmor.com>
Subject: Re: [Sip] verifying identity of sender
Date: Fri, 8 Oct 2004 06:42:26 -0500
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> Questions:
>
> 2) Is the benefit of the solution worth the additional 4 message 
> overhead on every transaction?

Dean/all -

2.a) is there anything that can be overlapped with this four-message 
exchange?

The calling party UA can immediately publish its hashed INVITE, and 
isn't waiting for the softarmor server's response to do anything - no 
delay there. But I'm thinking the called party UA couldn't advance to 
Alerting without getting the hashed response - am I missing anything?

I'm also thinking that an additional RTT from called party UA to 
calling party proxy would be more noticeable in wireless 
environments...

Spencer 



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Subject: Re: [Sip] Far End Camera Control in SIP
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On Oct 8, 2004, at 2:38 AM, Ranjit Avasarala wrote:

> Hi Rajeev
>
> what u say is ok. I mean u can indicate support for 224 in SDP. But IP 
> network is not capable of carrying delay sensitive information.
>
Ah, I didn't know that! Thanks, that'll save me a lot of work. We can 
shut down SIP, SIPPING, MMUSIC, AVT, and DIFFSERV and move on to other 
things.

dave.

> Ranjit
>
>
>
> Rajeew Kumar Singh wrote:
>> Hi Ranjit,
>>    Now people have started working on SIP based video conferencing.
>> What till now I came to know that the
>> "support for Far End Camera Control by SIP User Agent"
>>     can be signalled by SDP using the "m" field giving support for 
>> H.224
>> like
>> m:data  9000 RTP/AVP 100
>> Media Attribute(a):rtpmap:100 H224
>> This shows that the User Agent has support for Far End Camera Control.
>> Now the Far End Camera Control messages can be carried in RTP during 
>> the
>> conferencing.
>> The User agent should be intelligent enough to know the RTP streams 
>> related
>> to
>> Far End Camera Control..and accordingly the Camera is controlled.
>> Let me know if there is any misunderstanding on this part..
>> Thanks
>> Rajeew
>> -----Original Message-----
>> From: Ranjit Avasarala [mailto:ranjsadh@yahoo.com]
>> Sent: Thursday, October 07, 2004 7:56 PM
>> To: rajeewsingh@tataelxsi.co.in
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Far End Camera Control in SIP
>> Hi rajeev
>>  SIP is not the protocol of choice for far end camera
>> control or for video conferencing. More ideal protocol
>>  would be H.324 and H.324M for 3GPP.
>> Ranjit
>> --- Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
>> wrote:
>>> Hi,
>>> Can anybody provide some info on Far End Camera
>>> Control in SIP.
>>>
>>> Regards
>>> Rajeew
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list
>>> https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP
>>> Protocol
>>> Use sip-implementors@cs.columbia.edu for questions
>>> on current sip
>>> Use sipping@ietf.org for new developments on the
>>> application of sip
>>>
>> =====
>> Regards
>> Ranjit
>> _______________________________
>> Do you Yahoo!?
>> Declare Yourself - Register online to vote today!
>> http://vote.yahoo.com
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
> -- 
> Regards
> Ranjit
>
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0584901612==--




From sip-bounces@ietf.org  Fri Oct  8 10:40:28 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'IETF SIP Mailing List'" <sip@ietf.org>
Subject: FW: [Sip] Question on RFC3261 Merged Requests - please respond
Date: Fri, 8 Oct 2004 10:11:28 -0400
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Can someone - RFC3261 authors or otherwise - comment on this, please?=20

Regards, Shan Lu

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Shan
Lu
Sent: Friday, October 01, 2004 12:02 PM
To: 'Christer Holmberg (JO/LMF)'; 'IETF SIP Mailing List'
Subject: RE: [Sip] Question on RFC3261 Merged Requests


Christer,

I appreciate your comments. After thinking over the point you raised =
(why
old transaction mapping rules have to be used in 3261-compliant
environment), I came to the conclusion that 8.2.2.2 is completely =
broken.=20

My gateway example showed that at least when R-URI is different, second
INVITE should not be automatically rejected on the basis that it shares =
same
call-id, cseq, and the from tag with an INVITE in a different =
transaction.

Even when the two INVITEs have identical R-URI, the fact that they may =
have
visited different proxies after being forked could make one of them more
desirable than the other. The order they arrive at the UAS should not
determine which one gets presented to user and which one gets discarded. =
The
protocol can not make that distinction.

The forking proxy and calling UA both have built-in mechanisms to avoid =
one
single INVITE leading to two established dialogs. When it works, Section
8.2.2.2 is an optimization on top of that. And there are plenty of cases =
it
simply does not work.

I propose two options moving forward:

1. strike out 8.2.2.2 completely, or
2. replace it with a note to say that UAS may receive INVITEs in =
different
transactions but with identical call-id, cseq, from tag, and possibly =
same
R-URI. UAS should be prepared to handle this situation as it would with =
any
other incoming INVITEs.

I like to hear the list's opinion on this and whether it is a bug that
should be logged.

Regards,

Shan Lu
sentitO Networks =20

>-----Original Message-----
>From: Christer Holmberg (JO/LMF)=20
>[mailto:christer.holmberg@ericsson.com]=20
>Sent: Thursday, September 30, 2004 1:41 AM
>To: 'Shan Lu'; 'IETF SIP Mailing List'
>Subject: RE: [Sip] Question on RFC3261 Merged Requests
>
>
>
>Hi,
>
>Interesting point.
>
>One could argue that the existing text is ok for a normal SIP=20
>phone, because there is really no idea in receiving multiple=20
>"calls" (multiple "instances" of the same forked INVITE) from=20
>the same caller (in most cases the SIP phone can handle only=20
>one call at a time anyway, no matter if the incoming INVITEs=20
>origin from the same user or different users).
>
>However, for a gateway that may not be a valid case. Even if=20
>the gateway from a SIP perspective is a single SIP endpoint=20
>there are multiple end-users behind it, so IF we want to allow=20
>"forking" into the PSTN network (or whatever is on the other=20
>side) you are correct, I think.
>
>The problem I have with merged requests is that it takes away=20
>the advantages of only using the RFC3261 Via branch cookie=20
>from transaction identification. A node compliant with RFC3261=20
>will basically still have to use the "old" transaction mapping=20
>mechanism, in order the to the merge check...
>
>Regards,
>
>Christer Holmberg
>Ericsson Finland
>
>
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of
>> Shan Lu
>> Sent: 30. syyskuuta 2004 1:22
>> To: 'IETF SIP Mailing List'
>> Subject: [Sip] Question on RFC3261 Merged Requests
>>=20
>>=20
>> Hi,
>>=20
>> RFC3261 has following text:
>>=20
>> 8.2.2.2 Merged Requests
>>=20
>>    If the request has no tag in the To header field, the UAS=20
>core MUST
>>    check the request against ongoing transactions.  If the From tag,
>>    Call-ID, and CSeq exactly match those associated with an ongoing
>>    transaction, but the request does not match that=20
>transaction (based
>>    on the matching rules in Section 17.2.3), the UAS core SHOULD
>>    generate a 482 (Loop Detected) response and pass it to the server
>>    transaction.
>>=20
>> Assume that calls to support@example.com is forked to two=20
>> phone numbers on
>> PSTN, say, 555-1111 and 555-2222. Further assume that there=20
>> is only one
>> SIP-PSTN gateway. The two INVITEs arriving at the gw are=20
>> identical except
>> for R-URI userinfo and top VIA branch (assuming forking proxy=20
>> is the only
>> proxy).=20
>>=20
>> My question is: why should GW 482 second INVITE? Isn't that=20
>> the purpose of
>> forking? Should 8.2.2.2 be modified as:
>>=20
>> ... If request URI, the From tag, Call-ID, and CSeq exactly match ...
>>        ^^^^^^^^^^^^=20
>>=20
>> Regards,=20
>>=20
>> Shan Lu
>> sentitO Networks
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>=20
>


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Date: Fri, 08 Oct 2004 08:58:53 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sip] Session timer issue resolutions
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OK, I think its fair to characterize this as a typo, and I will change 
it to >= during auth48.

Thanks,
Jonathan R.

Ragunath Devarasu wrote:

> One reason could be backward compatibility with older drafts which said:
> 
> "                               If a Min-SE header is included
> in the initial session refresh request, the value of the
> Session-Expires MUST be equal to the value in Min-SE."
> 
> I understand that we needed to relax this requirement and that is the reason
> we have changed it to GREATER THAN. 
> 
> Assume that Session-Expires of already existing session is 300 seconds.
> During session timer refresh, uac wants to put Min-SE along with
> Session-Expires, it needs to put Min-SE: 299 and Session-Expires: 300 just
> to satisfy GREATER THAN. If it would have been GREATER THAN or EQUAL TO, uac
> could as well put Min-SE: 300 and Session-Expires: 300
> 
> Thanks
> Ragunath
> 
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
> Sent: Wednesday, September 15, 2004 8:43 PM
> To: Shriram Natarajan
> Cc: sip@ietf.org
> Subject: Re: [Sip] Session timer issue resolutions
> 
> I don't think there is really any difference between > and >= in this 
> case. Is there a specific reason you need equality?
> 
> Also, do note that session timer has been approved by IESG. As such, any 
> changes to this specification need to be done during authors 48 hours. 
> Such changes are meant to address editorial/typo problems, and bugs in 
> the spec. New features, improvements, etc. are not permitted.
> 
> Thanks,
> Jonathan R.
> 
> Shriram Natarajan wrote:
> 
> 
>>Good day Jonathan,
>>
>>Why must the initial session refresh request have SE > MinSE (if present)?
>>
>>draft-15 section 7.1 says:
>> >>
>>                            If a Min-SE header is included
>>   in the initial session refresh request, the value of the
>>   Session-Expires MUST be greater than the value in Min-SE.
>><<
>>
>>What is the problem in modifying this to say:
>>
>> >>
>>                            If a Min-SE header is included
>>   in the initial session refresh request, the value of the
>>   Session-Expires MUST be greater than or equal to the value
>>   in Min-SE.
>><<
>>
>>
>>With best regards
>>Shriram
>>
>>
>>
>>
>>On 7/18/2004 2:03 PM, Jonathan Rosenberg wrote:
>>
>>
>>>Folks,
>>>
>>>Over the past few months, there were a bunch of comments on the list 
>>>regarding the previous revision of session timer. That previous 
>>>revision, -13, was made to address IESG comments, as the document 
>>>completed working group last call and was making its way towards 
>>>approval.
>>>
>>>I'll summarize what was said and the conclusions that were reached.
>>>
>>>Comment: I never saw -13, what happened?
>>>
>>>Conclusion: There was some kind of secretariat problem, I don't know 
>>>why. It was resubmitted and a -14 appeared, with no content change 
>>>from -13.
>>>
>>>Comment: Making Min-SE a minimum of 90s is too low for applications 
>>>that want to know quickly when there is a failure, including billing.
>>>
>>>Conclusions: Usage of the session timer for such fine grained liveness 
>>>tests is simply out of scope. Session timer is explicitly designed to 
>>>deal with cleanup of old, stale state.
>>>
>>>There were some comments along the lines of, "don't impose this 90s 
>>>minimum because of philosophical objections to this usage". However, 
>>>the reason that the 90s minimum was introduces is not philosophical at 
>>>all. There was a REAL practical problem of sessions being 
>>>inappropriately timed out. Here is the example. We set up a call with 
>>>the minimum session timer. Half way through it, the client tries to 
>>>refresh. A few packets get lost, so the refresh takes a bit. However, 
>>>since the session timer was so small, the session gets torn down by 
>>>the peer UA and/or intermediaries. This is bad. The session should 
>>>continue as long as the refresh transaction can complete successfully, 
>>>if initiated halfway through the interval. Do a bit of math, and this 
>>>tells you that you need around a 90s minimum.
>>>
>>>A final point on this topic - I agree it would be useful for third 
>>>parties to get VERY fine grained notifications of session liveness and 
>>>failure. However, the session timer mechanism is just flat out 
>>>inappropriate for that. I think that, if this was the driving 
>>>requirement, you would define a totally different solution; something 
>>>that did very frequent, low-overhead messages (like a stun request for 
>>>example) directly between interested entitites. As such, if people 
>>>want a good way to solve this problem, please write some requirements 
>>>down, bring them forward, and we can work on it. But, session timer is 
>>>not going to EVER be good enough for 6s failure detections (6s was 
>>>explicitly mentioned in the thread as a goal).
>>>
>>>As such, I have not changed the 90s requirement.
>>>
>>>Comment: Paul made a comment, preferring the usage of "infinite" for a 
>>>UA and "indefinite" for a proxy, as a description of the default value 
>>>of the session timer (i.e., whats in place when nothing is 
>>>negotiated). Dean commented that IESG felt that this would not be 
>>>good, and that the document would have trouble passing with this change.
>>>
>>>Conclusion: The current wording stays.
>>>
>>>
>>>Comment: the document defines "initial session refresh" as a 
>>>mid-dialog request, but it can happen in the middle of a dialog.
>>>
>>>Conclusion: this is a good point. I have fixed the terminology.
>>>
>>>
>>>Comment: can session timer be used in SUBSCRIBE dialogs?
>>>
>>>Answer: I don't think its needed, since subscriptions have their own 
>>>way of refreshing. I see adding the session timer to that as 
>>>confusing. Granted, subscription durations aren't negotiated with 
>>>input from proxies, but I dont see why they'll care for the most part. 
>>>Invite sessions are different because of the heavy involvement from 
>>>intermediaries.
>>>
>>>Comment/Question: If UAC includes both Session-Expires and Min-SE in 
>>>an initial request, it is required to make the values the same 
>>>(section 7.1). Why???
>>>
>>>Answer:
>>>
>>>Good question. Turns out this is a bug, and there is a historical 
>>>reason for it.
>>>
>>>First, you must pay attention to this bit in the security 
>>>considerations section:
>>>
>>> Next, consider the case of a rogue UAS that wishes to force a UAC to
>>>   generate refreshes at a rapid rate. In that case, the UAC has to
>>>   support session timer. The initial INVITE arrives at the rogue UAS,
>>>   which returns a 2xx with a very small session interval. The UAC uses
>>>   this timer, and quickly sends a refresh. Section 7.1 requires the UAC
>>>   to copy the current session interval into the Session-Expires header
>>>   field in the request. This enables the proxies to see the current
>>>   value. The proxies will reject this request, and provide a Min-SE
>>>   with a higher minimum. The UAC will then use this higher minimum.
>>>   Note, that if the proxies did not reject the request, but rather
>>>   proxied the request with a Min-SE header field, an attack would still
>>>   be possible. The UAS could discard this header field in a 2xx
>>>   response, and force the UAC to continue to generate rapid requests.
>>>
>>> From this, it is clear that for subsequent refreshes (rather than 
>>>initial ones), the session-expires should be placed into the request 
>>>as it was, allowing the chance for proxies to reject it. Note, 
>>>however, that the text above references section 7.1 which talks about 
>>>*initial* session refresh requests. Now, in an earlier version of the 
>>>document, the description of initial and subsequent behavior was 
>>>combined, and the section reference was valid. After it was split 
>>>apart, the functionality was inappropriately left in the section on 
>>>initial requests, and the reference also stayed in tact.
>>>
>>>The requirement is there in a slightly different form for subsequent 
>>>session refreshes:
>>>
>>>   In a session refresh request sent within a dialog with an active
>>>   session timer, the Sesssion-Expires header field SHOULD be present.
>>>   When present, it MUST be equal to the maximum of the Min-SE header
>>>   field (recall that its default value when not present is 90 seconds)
>>>   and the current session interval.
>>>
>>>As such, I believe the constraint can be removed for initial requests. 
>>>I've fixed that (SE just has to be larger than Min-SE), and adjusted 
>>>the reference from the security considerations section.
>>>
>>>Now, the behavior for subsequent requests needs some further 
>>>consideration. Both strengths ought to be the same; there is no point 
>>>in making the Session Expires max(min,current) if its not going to be 
>>>present in the request. So, I changed the second MUST to a SHOULD, and 
>>>explained that you normally want this to avoid dos attacks, but can 
>>>remove it or set it larger if you want to "reset" the process and 
>>>compute a new Session expires.
>>>
>>>
>>>
>>>Comment: what if a proxy (such as that in an ALG) has a MAXIMUM timer 
>>>it can support? Can we reject a refresh request and include some kind 
>>>of max-se header field?
>>>
>>>Answer:
>>>
>>>You can't have it both ways. If you allow for rejection of requests 
>>>because an interval is above someones maximum, you can get a situation 
>>>where someones maximum is below someones minimum. This is unsolveable 
>>>by any protocol means - the constraint cannot be jointly met. So, 
>>>which is worse - having an interval thats longer than someone wants, 
>>>or shorter? I say, its worse to have one that is longer than someone 
>>>wants, because having one that is too short can generate a lot of 
>>>traffic and processing for everyone. Thus, for the purposes of being a 
>>>good sip citizen, it is better to have an interval be longer than you 
>>>might want.
>>>
>>>So, I have made no change to the spec.
>>>
>>>I'll post a separate note with the specific changes.
>>>
>>>-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: "Arunachalam Venkatraman \(arunvenk\)" <arunvenk@cisco.com>
To: "'Shan Lu'" <shanlu@sentito.com>, "'IETF SIP Mailing List'" <sip@ietf.org>
Subject: RE: [Sip] Question on RFC3261 Merged Requests
Date: Fri, 8 Oct 2004 09:53:12 -0500
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Shan
You are correct about this, In a gateway implementation I am familiar =
with,
the user part of the ReqUri are compared and if this also matches, a 482
response is sent to the second request. Otherwise, the second call is =
set up
to the distinctly different number.
Venkat


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Shan
Lu
Sent: Wednesday, September 29, 2004 5:22 PM
To: 'IETF SIP Mailing List'
Subject: [Sip] Question on RFC3261 Merged Requests


Hi,

RFC3261 has following text:

8.2.2.2 Merged Requests

   If the request has no tag in the To header field, the UAS core MUST
   check the request against ongoing transactions.  If the From tag,
   Call-ID, and CSeq exactly match those associated with an ongoing
   transaction, but the request does not match that transaction (based
   on the matching rules in Section 17.2.3), the UAS core SHOULD
   generate a 482 (Loop Detected) response and pass it to the server
   transaction.

Assume that calls to support@example.com is forked to two phone numbers =
on
PSTN, say, 555-1111 and 555-2222. Further assume that there is only one
SIP-PSTN gateway. The two INVITEs arriving at the gw are identical =
except
for R-URI userinfo and top VIA branch (assuming forking proxy is the =
only
proxy).=20

My question is: why should GW 482 second INVITE? Isn't that the purpose =
of
forking? Should 8.2.2.2 be modified as:

... If request URI, the From tag, Call-ID, and CSeq exactly match ...
       ^^^^^^^^^^^^=20

Regards,=20

Shan Lu
sentitO Networks


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Subject: Re: [Sip] verifying identity of sender
From: Aki Niemi <aki.niemi@nokia.com>
To: ext Dean Willis <dean.willis@softarmor.com>
In-Reply-To: <4165B3CF.8020707@softarmor.com>
References: <BD8AB5BD.143BE%fluffy@cisco.com> <4165B3CF.8020707@softarmor.com>
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On Fri, 2004-10-08 at 00:23, ext Dean Willis wrote:
...snip...
> I INVITE you to a call.  Upon sending the INVITE, my UA sends a PUBLISH 
> to the softarmor.com server saying I really did send an INVITE to you, 
> and including a hash of that INVITE. Hopefully, softarmor.com 
> authenticates that PUBLISH message.

Uhh... I don't really want to nit-pick on this proposal, but this isn't
the intended use of the PUBLISH method [1].

I guess if your UA sent the INVITE first to softarmor.com's outbound
proxy, who in turn Digest-challenged for your proxy-authentication, then
no PUBLISH would be needed.

Cheers,
Aki

[1] From Abstract:

   ...
   The mechanism described in this document can be extended to support
   publication of any event state for which there exists an appropriate
   event package.  It is not intended to be a general-purpose mechanism
   for transport of arbitrary data, as there are better-suited
   mechanisms for this purpose.

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Date: Fri, 08 Oct 2004 11:28:48 -0700
Subject: Re: [Sip] verifying identity of sender
From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>
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ok, ignoring the syntax of the message, what you have is A and B are proxies
and A sends message to B with a token in it then B contacts A and verifies
this. This works as well as B can believe that it reached A and not the
attacker. 

Now consider the current identity proposal when A uses a self signed cert.
A adds hash and sends message to B. B contacts A to get the certificate. If
B decided to accept this certificate from someone that provides a self
signed cert it can. This is just policy about how well it wants to believe
it really is talking to B. I think that in this case, the identity scheme
reduces to exactly the same basic message flow and security properties as
your proposal below.



On 10/7/04 2:23 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> Cullen Jennings wrote:
> 
>>> here nothing new needs to be put in dns.  when uas receives invite, it
>>> gives back a provisional response, constructs the verify request
>>> including a hash of invite's from uri, call id and from tag, and then
>>> sends it out like any out-of-dialog request using the from uri as
>>> request uri.  when the uac of the invite receives the verify request, it
>>> checks if it has any pending invite that matches the hash and responds
>>> either with 200 ok or some failure code.  when uas receives the verify
>>> response it gives final response to the invite and that is it.
>>> 
>>>    
>>> 
>> 
>> ok so my from says if X send you a message, then X can receive your reply.
>> How does this help know who X is?
>>  
>> 
> Not quite.
> 
> I INVITE you to a call.  Upon sending the INVITE, my UA sends a PUBLISH
> to the softarmor.com server saying I really did send an INVITE to you,
> and including a hash of that INVITE. Hopefully, softarmor.com
> authenticates that PUBLISH message.
> 
> You get the invite, wonder if  it really came from someone who could
> authenticate as me to softarmor.com, poll that server (including the
> hash) and softarmor.com tells you "Yes, that INVITE came from somebody
> who could authenticate as Dean. Trust me."
> 
> So, the strength of the assertion of softarmor.com as to my identity is
> contingent on two things:
> 
> 1) The strength of your belief that the server you think is
> softarmor.com really is (and isn't somebody else just faking)
> 2) The strength of your belief in the functional integrity and
> authentication provided by softarmor.com
> 
> In the simplest case, all that this technique proves is that whoever
> sent you the message is either 1)  authenticated to the asserting
> server, or 2) capable of munging DNS or IP routing to intercept your
> request to the asserting server, making them sufficiently interesting to
> talk to anyhow.
> 
> In the typical "dumb ISP" case, this is a weak assertion, but it's still
> much stronger than we get for email and would block the majority of
> source-fraud spam.
> 
> But in a confederated environment, where servers might have IPSEC SAs or
> be linked by private, protected network, the assertion could be a bit
> stronger.
> 
> 
> Questions:
> 
> 1) Is this imperfect solution so useful as to be worth specifying, even
> though we know it's not complete?
> 
> 2) Is the benefit of the solution worth the additional 4 message
> overhead on every transaction?
> 
> 3) Are there obvious  optimizations, such as having the asserting server
> "learn" about the request by observation rather than explicit notification?
> 
> 4) Could we just use a MARID approach, and configure "edge" proxies to
> discard inbound requests where the source IP address doesn't show in DNS
> as authoritative for the domain component of the From: address, thereby
> getting most of the benefit with less overhead?
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From: "Even, Roni" <roni.even@polycom.co.il>
To: rajeewsingh@tataelxsi.co.in, sip@ietf.org
Subject: RE: [Sip] Far End Camera Control in SIP
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------_=_NextPart_000_01C4AD67.C78362EF
Content-Type: text/plain

Hi 
I just submitted a personal draft to the AVT about FECC. We try to see which
working group item it should be
Roni Even

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Rajeew
Kumar Singh
Sent: Wednesday, October 06, 2004 2:11 PM
To: sip@ietf.org
Subject: [Sip] Far End Camera Control in SIP


Hi,
 Can anybody provide some info on Far End Camera Control in SIP.

Regards
Rajeew


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AVT                                                              R. =
Even
Internet-Draft                                                   =
Polycom
Expires: March 2, 2005                                    September =
2004


                  Far End Camera Control Payload Type
                       draft-even-avt-h224-00.txt

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of section 3 of RFC 3667.  By submitting this Internet-Draft, each
   author represents that any applicable patent or other IPR claims of
   which he or she is aware have been or will be disclosed, and any of
   which he or she become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as
   Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six =
months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on March 2, 2005.

Copyright Notice

   Copyright (C) The Internet Society (2004).

Abstract

   This document defines the syntax and the semantics of SDP parameters
   needed to support far end camera control protocol.  In conversional
   video applications far end camera control protocol is used by
   participants to control the remote camera.  The common used protocol
   is ITU H.281 over H.224.






Even                     Expires March 2, 2005                  [Page =
1]
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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  =
3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  =
4
   3.  Far-end camera control protocol  . . . . . . . . . . . . . . .  =
5
   4.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  =
6
     4.1   Registration of MIME media type application/h224 . . . . .  =
6
   5.  SDP Parameters . . . . . . . . . . . . . . . . . . . . . . . .  =
7
     5.1   Usage with the SDP Offer Answer Model  . . . . . . . . . .  =
7
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . .  =
8
   7.  Normative References . . . . . . . . . . . . . . . . . . . . .  =
8
       Author's Address . . . . . . . . . . . . . . . . . . . . . . .  =
9
       Intellectual Property and Copyright Statements . . . . . . . . =
10






































Even                     Expires March 2, 2005                  [Page =
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2004


1.  Introduction

   The ITU-T recommendation H.281 [ITU.281] specifies a protocol for =
far
   end camera control.  This protocol is carried in H.320 systems using
   H.224 [ITU.H224] H.323 annex Q specifies how to carry H.281/H.224
   frames using RTP packets.

   The document will define the SDP[RFC2327]parameters needed to =
support
   the above far end camera control protocol in systems that use SDP










































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2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC2119[RFC2119] and
   indicate requirement levels for compliant RTP implementations.













































Even                     Expires March 2, 2005                  [Page =
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2004


3.  Far-end camera control protocol

   This protocol is based on ITU-T H.281 running over ITU-T H.224 in an
   RTP/UDP channel.  H.323 annex Q specifies how to build the RTP
   packets from the H.224 packets.

   Using far end camera control protocol in point to point calls and
   multipoint calls is described in H.281 and H.323 annex Q











































Even                     Expires March 2, 2005                  [Page =
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4.  IANA Considerations

   This section describes the MIME types and names associated with this
   payload format.  The section registers the MIME types, as per
   RFC2048[RFC2048]

4.1  Registration of MIME media type application/h224

   MIME media type name: application

   MIME subtype name: H224

   Required parameters: None

   Optional parameters:  None

   Encoding considerations:

   This type is only defined for transfer via RTP [RFC3550]

   Security considerations: See Section 6

   Interoperability considerations: Terminals that will like to send =
far
   end camera control command should use this MIME type, receivers who
   can not support the protocol will reject the channel.

   Published specification: RFC yyy

   Applications which use this media type:

   Video conferencing applications.

   Additional information: none

   Person and email address to contact for further information :

   Roni Even: roni.even@polycom.co.il

   Intended usage: COMMON

   Author/Change controller:

   Roni Even








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5.  SDP Parameters

   The MIME media type application/h224 string is mapped to fields in
   the Session Description Protocol (SDP)  as follows:

   o The media name in the "m=3D" line of SDP MUST be application.  The
   transport SHOULD be RTP and the payload type is dynamic.

   o The encoding name in the "a=3Drtpmap" line of SDP MUST be h224 =
(the
   MIME subtype).

   o The clock rate in the "a=3Drtpmap" line MUST be 0.

   The recommended maximum bandwidth for this protocol is 6.4 kbit/sec.

5.1  Usage with the SDP Offer Answer Model

   When offering FECC using SDP in an Offer/Answer model[RFC3264]  the
   following considerations are necessary.

   H.281 Far end camera control communication are uni-directional.
   H.224 is bi-directional and can be used to learn the capabilities of
   the remote video end point e.g how many cameras it has.  The offer
   answer exchange will be dependent on the functionality of both side.

   The offerer will offer a sendonly channel if its camera can not be
   remotely controlled and if the offerer does not intend to use H.224
   to learn the capabilities of the remote video endpoints.

   In all other cases, when the offerer camera can be remotely
   controlled and/or it intends to use H.224 capabilities negotiation,
   the offerer will offer a sendrecv channel.

   The answerer behavior will be as follows:

   If it receives an offer with sendonly it will answer with a recvonly
   if it supports far end camera control, otherwise it will ignore
   reject the offer.

   If it receives an offer with sendrecv and its camera can be remotely
   controlled it will answer with a sendrecv option.  If its camera
   cannot be remotely control it will reject the offer but may later =
try
   to remotely control the offerer's camera using this procedure.








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6.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550].  This implies that confidentiality of the
   media streams is achieved by encryption.

   A potential denial-of-service threat exists.  The attacker can =
inject
   pathological datagrams  into the stream which may cause the receiver
   to move the camera randomly.  The usage of authentication of at =
least
   the RTP packet is RECOMMENDED

   As with any IP-based protocol, in some circumstances a receiver may
   be overloaded simply by the receipt of too many packets, either
   desired or undesired.  Network-layer authentication may be used to
   discard packets from undesired sources, but the processing cost of
   the authentication itself may be too high.

7  Normative References

   [ITU.281]  International Telecommunications Union, "A far end camera
              control protocol for videoconferences using H.224", ITU-T
              Recommendation H.281, November 1994.

   [ITU.H224]
              International Telecommunications Union, "A real time
              control protocol for simplex applications using the H.221
              LSD/HSD/HLP channels.", ITU-T Recommendation H.224,
              February 2000.

   [ITU.H323]
              International Telecommunications Union, "Visual telephone
              systems and equipment for local area networks which
              provide a non-guaranteed quality of service", ITU-T
              Recommendation H.323, July 2003.

   [RFC2048]  Freed, N., Klensin, J. and J. Postel, "Multipurpose
              Internet Mail Extensions (MIME) Part Four: Registration
              Procedures", BCP 13, RFC 2048, November 1996.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2327]  Handley, M. and V. Jacobson, "SDP: Session Description
              Protocol", RFC 2327, April 1998.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June



Even                     Expires March 2, 2005                  [Page =
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              2002.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R. and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.


Author's Address

   Roni Even
   Polycom
   94 Derech Em Hamoshavot
   Petach Tikva  49130
   Israel

   EMail: roni.even@polycom.co.il



































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2004


Intellectual Property Statement

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Even                     Expires March 2, 2005                 [Page =
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=0C

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
------_=_NextPart_000_01C4AD67.C78362EF--



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From: Dean Willis <dean.willis@softarmor.com>
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To: Aki Niemi <aki.niemi@nokia.com>
Subject: Re: [Sip] verifying identity of sender
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Aki Niemi wrote:
> On Fri, 2004-10-08 at 00:23, ext Dean Willis wrote:
> ...snip...
> 
>>I INVITE you to a call.  Upon sending the INVITE, my UA sends a PUBLISH 
>>to the softarmor.com server saying I really did send an INVITE to you, 
>>and including a hash of that INVITE. Hopefully, softarmor.com 
>>authenticates that PUBLISH message.
> 
> 
> Uhh... I don't really want to nit-pick on this proposal, but this isn't
> the intended use of the PUBLISH method [1].

Absolutely true.

> I guess if your UA sent the INVITE first to softarmor.com's outbound
> proxy, who in turn Digest-challenged for your proxy-authentication, then
> no PUBLISH would be needed.

yes, that's true.

> [1] From Abstract:
> 
>    ...
>    The mechanism described in this document can be extended to support
>    publication of any event state for which there exists an appropriate
>    event package.  It is not intended to be a general-purpose mechanism
>    for transport of arbitrary data, as there are better-suited
>    mechanisms for this purpose.

of course, one could certainly define an event package for "Pending 
Requests for Which I Might Need a Half-Assed Identity Voucher". Perhaps 
a modification of the UA State event package?

This does give rise to the question of "How do we evaluate the 
appropriateness of new event packages?" which I suspect we'll be asked 
to do over the next year or so. Something to think about, as many of the 
proposals I've heard for new event packages had even less merit than the 
strawman above.

--
Dean


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From sip-bounces@ietf.org  Mon Oct 11 00:27:34 2004
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To: David R Oran <oran@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Far End Camera Control in SIP
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Hi
      What I meant by "IP network is not capable" is still IP network 
has to evolve and lot of work needs to be done to make it robustto carry 
delay sensitive data ..Thats why SIP, SIPPING, SIMPLE , etc groups are 
there..

Dave, Hope u have got what I meant.

Ranjit

David R Oran wrote:
> 
> On Oct 8, 2004, at 2:38 AM, Ranjit Avasarala wrote:
> 
>> Hi Rajeev
>>
>> what u say is ok. I mean u can indicate support for 224 in SDP. But IP 
>> network is not capable of carrying delay sensitive information.
>>
> Ah, I didn't know that! Thanks, that'll save me a lot of work. We can 
> shut down SIP, SIPPING, MMUSIC, AVT, and DIFFSERV and move on to other 
> things.
> 
> dave.
> 
>> Ranjit
>>
>>
>>
>> Rajeew Kumar Singh wrote:
>>
>>> Hi Ranjit,
>>>    Now people have started working on SIP based video conferencing.
>>> What till now I came to know that the
>>> "support for Far End Camera Control by SIP User Agent"
>>>     can be signalled by SDP using the "m" field giving support for H.224
>>> like
>>> m:data  9000 RTP/AVP 100
>>> Media Attribute(a):rtpmap:100 H224
>>> This shows that the User Agent has support for Far End Camera Control.
>>> Now the Far End Camera Control messages can be carried in RTP during the
>>> conferencing.
>>> The User agent should be intelligent enough to know the RTP streams 
>>> related
>>> to
>>> Far End Camera Control..and accordingly the Camera is controlled.
>>> Let me know if there is any misunderstanding on this part..
>>> Thanks
>>> Rajeew
>>> -----Original Message-----
>>> From: Ranjit Avasarala [mailto:ranjsadh@yahoo.com]
>>> Sent: Thursday, October 07, 2004 7:56 PM
>>> To: rajeewsingh@tataelxsi.co.in
>>> Cc: sip@ietf.org
>>> Subject: Re: [Sip] Far End Camera Control in SIP
>>> Hi rajeev
>>>  SIP is not the protocol of choice for far end camera
>>> control or for video conferencing. More ideal protocol
>>>  would be H.324 and H.324M for 3GPP.
>>> Ranjit
>>> --- Rajeew Kumar Singh <rajeewsingh@tataelxsi.co.in>
>>> wrote:
>>>
>>>> Hi,
>>>> Can anybody provide some info on Far End Camera
>>>> Control in SIP.
>>>>
>>>> Regards
>>>> Rajeew
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list
>>>> https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP
>>>> Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions
>>>> on current sip
>>>> Use sipping@ietf.org for new developments on the
>>>> application of sip
>>>>
>>> =====
>>> Regards
>>> Ranjit
>>> _______________________________
>>> Do you Yahoo!?
>>> Declare Yourself - Register online to vote today!
>>> http://vote.yahoo.com
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>
>>
>>
>> -- 
>> Regards
>> Ranjit
>>
>>
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com


-- 
Regards
Ranjit





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From sip-bounces@ietf.org  Mon Oct 11 00:31:18 2004
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Cullen Jennings wrote:
> ok, ignoring the syntax of the message, what you have is A and B are proxies
> and A sends message to B with a token in it then B contacts A and verifies
> this. This works as well as B can believe that it reached A and not the
> attacker. 


A simple two-party reverse validity check can be handled more readily 
with a null-poassword digest authentication request, as proposed on the 
list a few years back.

Actually, you need 3 parties to demonstrate the utility of Juha's proposal.

Assume A, B, and C where A and C are UAs, and B is a proxy normally used 
by neither, but authoritative for the domain of A.

A sends a message to C with a token in it, which C can then ask B to 
validate -- "Hey Dude, do you believe one of your people called A said 
this to me?"


> Now consider the current identity proposal when A uses a self signed cert.
> A adds hash and sends message to B. B contacts A to get the certificate. If
> B decided to accept this certificate from someone that provides a self
> signed cert it can. This is just policy about how well it wants to believe
> it really is talking to B. I think that in this case, the identity scheme
> reduces to exactly the same basic message flow and security properties as
> your proposal below.

err, not exactly sure I follow the above, as you seem to have B using 
A's cert to decide whether it is talking to itself.

But the (relatively) important thing about this approach is that it does 
block attacks where somebody pretends to be A, but doesn't really have 
the wherewithal to either authenticate to B, or to subvert the mostly 
unprotected the B-C link. This is excatly the weakness that the majority 
of source-forged spammers use to deliver spam in today's world.


--
Dean

--
dean

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Dean,

Help me to understand your position here. Is it your position as WG chair
that we need to scrap the existing identity work, and begin anew along the
lines you describe below? Or would you like to pursue this work as a
separate work item?

Anyway, hypothetically exploring this line of thought, it seems that using
PUBLISH to inform your local server of the hash of the INVTE you've sent is
probably superfluous - why not just send the INVITE through the local
server, and have it act as a proxy (along the lines of Question 3 in your
original mail describing the scheme)? It can acquire whatever state it needs
to as the request passes (including generating or copying the hash).

If you don't do that, anyway, you prevent the domain from enforcing any
identity policies on requests before they are sent. While this isn't
strictly necessary for an identity system to operate, it does provide a
number of beneficial properties.

If you do just sent the INVITE through the local server, the only real
difference between sip-identity-03 and your approach is whether the local
server signs the hash with a cert or forces any verifier to do a dial-back.
In fact, this variety of dial-back is remarkably similar to the
authentication mechanism employed by Jabber. That much said, sip-identity-03
can also requires a dial-back, after a fashion, if the verifier does not
already have the cert of the proxy server.

So, once again, this is a question about how to provide a construct that
links a domain name to some sort of credentials: cryptographically with
certificates, or using the network with the DNS.

A couple of things to note, beyond the general weaknesses of using the DNS
already highlighted in this thread:

- There can be more than one consumer of the identity information in a
request; i.e., one or more intermediaries and/or the UAS might want to
verify the hash. If a dial-back is required, we pay the network cost on a
per-verifier basis. Of course, in sip-identity-03, if verifiers don't have
the cert of the domain, they'll pay the network cost anyway. However, each
verifier needs to pay that network cost only once per unique domain, not
once per unique message. This can quickly become meaningful, even in the
course of a single dialog.

- In order to dial-back (not to mention PUBLISH) securely, one should use
TLS. TLS entails the use of certs anyway, and requires the verifier to
download a cert. So, it's not clear to me that the cert costs won't be paid
anyway. Without TLS, anyway, you have to worry about more than just someone
who is capable of munging DNS or IP routing - passive attackers could quite
possibly confuse the verifier with forged responses to a poll to verify the
hash.

- This requires per-message state (i.e. the hash) to be maintained at the
local server (as your proposed use of PUBLISH exemplifies). Managing an
event database that changes for each message in your domain becomes
intensive pretty quick. Of course, sip-identity-03 requires state to be kept
at verifiers, in the form of certificates, but this is not per-message
state, nor even per-sender state - it's per-sending-domain state. Moreover,
for the most part this state is being kept where state belongs - in the UAs,
not in intermediaries.

Finally, let me add that the dial-back constraint in sip-identity-03
(deferencing the URI in the Identity-Info header to acquire the cert) is
artificial - it's not necessary for this mechanism use the network. The
originating UAC could of course include his domain's certificate in a
request with a "cid:" URI in the Identity-Info header indicating the MIME
body contain the cert. The UAC will know this cert, after all, because at
the very least it will learn it when it forges a TLS connection to the
authentication service. If we did this every time, there would never be any
need for the verifier to poll the network. Of course, it would also make
requests larger, and this is why the draft currently doesn't recommend the
practice. But the fact that you can do things this way without impacting the
strength of the solution shows, I think, the versatility of this approach.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Sunday, October 10, 2004 9:25 PM
> To: Cullen Jennings
> Cc: sip@ietf.org; Juha Heinanen
> Subject: Re: [Sip] verifying identity of sender
> 
> 
> Cullen Jennings wrote:
> > ok, ignoring the syntax of the message, what you have is A 
> and B are proxies
> > and A sends message to B with a token in it then B contacts 
> A and verifies
> > this. This works as well as B can believe that it reached A 
> and not the
> > attacker. 
> 
> 
> A simple two-party reverse validity check can be handled more readily 
> with a null-poassword digest authentication request, as 
> proposed on the 
> list a few years back.
> 
> Actually, you need 3 parties to demonstrate the utility of 
> Juha's proposal.
> 
> Assume A, B, and C where A and C are UAs, and B is a proxy 
> normally used 
> by neither, but authoritative for the domain of A.
> 
> A sends a message to C with a token in it, which C can then ask B to 
> validate -- "Hey Dude, do you believe one of your people 
> called A said 
> this to me?"
> 
> 
> > Now consider the current identity proposal when A uses a 
> self signed cert.
> > A adds hash and sends message to B. B contacts A to get the 
> certificate. If
> > B decided to accept this certificate from someone that 
> provides a self
> > signed cert it can. This is just policy about how well it 
> wants to believe
> > it really is talking to B. I think that in this case, the 
> identity scheme
> > reduces to exactly the same basic message flow and security 
> properties as
> > your proposal below.
> 
> err, not exactly sure I follow the above, as you seem to have B using 
> A's cert to decide whether it is talking to itself.
> 
> But the (relatively) important thing about this approach is 
> that it does 
> block attacks where somebody pretends to be A, but doesn't 
> really have 
> the wherewithal to either authenticate to B, or to subvert the mostly 
> unprotected the B-C link. This is excatly the weakness that 
> the majority 
> of source-forged spammers use to deliver spam in today's world.
> 
> 
> --
> Dean
> 
> --
> dean
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Date: Mon, 11 Oct 2004 16:08:08 +0200
Subject: Re: [Sip] verifying identity of sender
From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>
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inline ...

On 10/11/04 6:24 AM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> Cullen Jennings wrote:
>> ok, ignoring the syntax of the message, what you have is A and B are proxies
>> and A sends message to B with a token in it then B contacts A and verifies
>> this. This works as well as B can believe that it reached A and not the
>> attacker. 
> 
> 
> A simple two-party reverse validity check can be handled more readily
> with a null-poassword digest authentication request, as proposed on the
> list a few years back.

Hmm, I don't by the "handled more readily" part. I have looked at dial back
and I believe it is harder to implement, results in weaker security in all
cases, is not easier to deploy, and results in significantly higher load on
proxies. I know that at first glance it seems like anything that does not
involve PKI must be easier than anything that does but  I don't think it is
in this case. I will try and explain why later on in this email.
 
> 
> Actually, you need 3 parties to demonstrate the utility of Juha's proposal.
> 
> Assume A, B, and C where A and C are UAs, and B is a proxy normally used
> by neither, but authoritative for the domain of A.
> 
> A sends a message to C with a token in it, which C can then ask B to
> validate -- "Hey Dude, do you believe one of your people called A said
> this to me?"
> 

Hmm suspect you often need another proxy that is authoritative for B to form
the dialog but never mind.
 
So the dial back proposal works something like  A first gets this token, on
a per call basis from B (it has to be per call to include enough in the
token to stop cut and paste attacks). So B gets token, and hands to C. C
does connection with B and verifies the token. Is that connection TLS
protected? 

Alternative call flow using the Identity mechanism falls into two cases, one
where B trusts A with B's private key. This would be the case for voice mail
systems, large gateways, conference servers, and such but not typically the
case for phones other than in domains with a very small number of phones.
Keep in mind this thread started around domains that were too cheap to get a
certificate from a well known CA.

If B trusts A, then A can insert the identity header, send the call to C. If
the cert is from a well known CA then CA then C can verify the identity of
the sender. Note number of messages involved was 1.

If B does not trust A, then A can send call to B using previous digest
credential and B can insert identity header and send call to C and C can
verify it. Message count 2.

If B does not have a cert from a well known CA, it can still do the same
things as before but C will need to fetch the cert from B. The *first* time
this happens we add a +2 to the message count but it can be cached for later
messages. 


> 
>> Now consider the current identity proposal when A uses a self signed cert.
>> A adds hash and sends message to B. B contacts A to get the certificate. If
>> B decided to accept this certificate from someone that provides a self
>> signed cert it can. This is just policy about how well it wants to believe
>> it really is talking to B. I think that in this case, the identity scheme
>> reduces to exactly the same basic message flow and security properties as
>> your proposal below.
> 
> err, not exactly sure I follow the above, as you seem to have B using
> A's cert to decide whether it is talking to itself.
>

Hopefully I cleared this up somewhat in the example of above. If this is not
clear, then is is probably not clear to many people and we should spend some
time at the next IETF meeting discussing why it is unclear and how to get it
clear.  

> But the (relatively) important thing about this approach is that it does
> block attacks where somebody pretends to be A, but doesn't really have
> the wherewithal to either authenticate to B, or to subvert the mostly
> unprotected the B-C link. This is excatly the weakness that the majority
> of source-forged spammers use to deliver spam in today's world.
> 

You are correct that the proxy to proxy link is were most SPAM gets inserted
today but DNS dialback is not going to help this all that much (unless you
assume DNSSec). That is why the email folks are so interested in the MASS
working group which ends up with solutions very similar to SIP Identity.

As a side note on DNS Sec - some people feel that PKI with delegated signing
keys for every company failed because the big CAs were greedy. Perhaps they
were not greed and that's just what it costs to run a CA - either way my
argument stays the same. DNS Sec requires the people that run the top level
domains to delegate to the sub domains and not be too greedy. Unfortunately,
the companies that ran the CAs and the companies that will with run the DNS
Sec TLDs are, surprise, surprise, the same companies. They may change their
pricing but I believe they are just as likely to change their pricing on
delegated CA certs as they are on sub domain signing)

Dialback provides better security than nothing and does not require money to
be sent to any CAs. The Identity draft as written provides a way to
implement one thing and be able to simultaneously get a much better solution
that does use CAs and also be able to get a solution that is better than
Dialback and also does not require certificates from well known CA's.

The level of security offered by Dialback is ok for some. Mostly for people
that are willing to use no security but would like better as long as it cost
them nothing. Identity can provide this (including the no extra cost part)

People that need security better than none, usually need security well
beyond what dialback can provide. Identity can provide that too.

There is a large contingent of people that want real security available.
This includes encrypted integrity protect communications to the person you
think you are talking to. It's useless to have encrypted communications but
not know who it is encrypted to. Identity meets the needs of knowing who you
are talking to for this group of people and is much more deployable than a
S/MIME cert for every UA.

It's not clear to me what problem dialback solves that many deployments
would care about. Today the B-C links are very well protected as they mostly
run in IPSec tunnels between preconfigured links. The ones that are not all
preconfigured can use mutual TLS.




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The IESG has received a request from the Session Initiation Protocol WG to 
consider the following document:

- 'Update to the Session Initiation Protocol (SIP) Preconditions Framework '
   <draft-ietf-sip-rfc3312-update-03.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2004-10-25.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-rfc3312-update-03.txt


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Peterson, Jon wrote:

>Dean,
>
>Help me to understand your position here. Is it your position as WG chair
>that we need to scrap the existing identity work, and begin anew along the
>lines you describe below? Or would you like to pursue this work as a
>separate work item?
>
>  
>

My position is that exploring Juha's suggested approach through list 
dicussion may provide enlightenment. I have not taken a position as 
chair on the appropriateness of this work.

>Anyway, hypothetically exploring this line of thought, it seems that using
>PUBLISH to inform your local server of the hash of the INVTE you've sent is
>probably superfluous - why not just send the INVITE through the local
>server, and have it act as a proxy (along the lines of Question 3 in your
>original mail describing the scheme)? It can acquire whatever state it needs
>to as the request passes (including generating or copying the hash).
>
>  
>
This was previously suggested by Aki, and is inarguably true. It does, 
however, predicate sending the request through the proxy, which is not 
absolutely required to get the properties under discussion. In short, 
it's a useful optimization, but is no more revealing (and may, in fact 
obfuscate) the security properties being discussed in Juha's proposal, 
as I understand them.  This approach is roughly analogous to the 
lightweight alternative I proposed earlier, which is "accept no requests 
from an external domain unless they came directly from the IP address of 
a proxy that is authoritative for that domain according to DNS". Again, 
this is not perfect security, but if combined with TCP (or self-signed 
TLS) at least blocks the vast majority of source-addres forgeries as 
seen in the world of email.

>If you don't do that, anyway, you prevent the domain from enforcing any
>identity policies on requests before they are sent. While this isn't
>strictly necessary for an identity system to operate, it does provide a
>number of beneficial properties.
>
>  
>
Right. These properties were explicitly not part of Juha's proposal, so 
I constructed a discussion framework that did not exhibit them.

>If you do just sent the INVITE through the local server, the only real
>difference between sip-identity-03 and your approach is whether the local
>server signs the hash with a cert or forces any verifier to do a dial-back.
>In fact, this variety of dial-back is remarkably similar to the
>authentication mechanism employed by Jabber. That much said, sip-identity-03
>can also requires a dial-back, after a fashion, if the verifier does not
>already have the cert of the proxy server.
>
>  
>
I believe that is correct.

>So, once again, this is a question about how to provide a construct that
>links a domain name to some sort of credentials: cryptographically with
>certificates, or using the network with the DNS.
>
>A couple of things to note, beyond the general weaknesses of using the DNS
>already highlighted in this thread:
>
>- There can be more than one consumer of the identity information in a
>request; i.e., one or more intermediaries and/or the UAS might want to
>verify the hash. If a dial-back is required, we pay the network cost on a
>per-verifier basis. Of course, in sip-identity-03, if verifiers don't have
>the cert of the domain, they'll pay the network cost anyway. However, each
>verifier needs to pay that network cost only once per unique domain, not
>once per unique message. This can quickly become meaningful, even in the
>course of a single dialog.
>
>  
>
Quite true, and a strong argument indeed.

>- In order to dial-back (not to mention PUBLISH) securely, one should use
>TLS. TLS entails the use of certs anyway, and requires the verifier to
>download a cert. So, it's not clear to me that the cert costs won't be paid
>anyway. Without TLS, anyway, you have to worry about more than just someone
>who is capable of munging DNS or IP routing - passive attackers could quite
>possibly confuse the verifier with forged responses to a poll to verify the
>hash.
>
>  
>
That depends to a great extent on your definition of "secure". There is, 
in a practical sense, a significant difference between state-of-the-art 
(or state of the imagination, in some cases) security and "things that 
can be done very esaily and help prevent real problems." Since someone 
with paractical experience running a consumer network raised the 
question along with the argument that the identity approach might be 
difficult for them to deploy, it seems like it might be worth discussing 
the issues.

Or in short, "Security doesn't have to be perfect to be useful".

So, does Juha's approach offer any security benefits over simpler hacks, 
like my source-to-authority matching requirement? If it does, then let's 
understand those properties and see if any sort of standardization is 
needed. If not, then let's see what properties really are useful, and 
see if we need to document practices that exploit those properties for 
the operator community.

>- This requires per-message state (i.e. the hash) to be maintained at the
>local server (as your proposed use of PUBLISH exemplifies). Managing an
>event database that changes for each message in your domain becomes
>intensive pretty quick. Of course, sip-identity-03 requires state to be kept
>at verifiers, in the form of certificates, but this is not per-message
>state, nor even per-sender state - it's per-sending-domain state. Moreover,
>for the most part this state is being kept where state belongs - in the UAs,
>not in intermediaries.
>  
>
Absolutely true, and again a very strong argument.

>Finally, let me add that the dial-back constraint in sip-identity-03
>(deferencing the URI in the Identity-Info header to acquire the cert) is
>artificial - it's not necessary for this mechanism use the network. The
>originating UAC could of course include his domain's certificate in a
>request with a "cid:" URI in the Identity-Info header indicating the MIME
>body contain the cert. The UAC will know this cert, after all, because at
>the very least it will learn it when it forges a TLS connection to the
>authentication service. If we did this every time, there would never be any
>need for the verifier to poll the network. Of course, it would also make
>requests larger, and this is why the draft currently doesn't recommend the
>practice. But the fact that you can do things this way without impacting the
>strength of the solution shows, I think, the versatility of this approach.
>  
>
All true. But as Juha said, this approach requires gettting a valid, 
signed, checkable cert, which is non-trivial. I've been running my own 
fake CA for years, and I don't have any "real" certs to use. Do you?

--
Dean

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From: Dean Willis <dean.willis@softarmor.com>
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Cullen Jennings wrote:

>
>>Actually, you need 3 parties to demonstrate the utility of Juha's proposal.
>>
>>Assume A, B, and C where A and C are UAs, and B is a proxy normally used
>>by neither, but authoritative for the domain of A.
>>
>>A sends a message to C with a token in it, which C can then ask B to
>>validate -- "Hey Dude, do you believe one of your people called A said
>>this to me?"
>>
>>    
>>
>
>Hmm suspect you often need another proxy that is authoritative for B to form
>the dialog but never mind.
> 
>So the dial back proposal works something like  A first gets this token, on
>a per call basis from B (it has to be per call to include enough in the
>token to stop cut and paste attacks). So B gets token, and hands to C. C
>does connection with B and verifies the token. Is that connection TLS
>protected? 
>
>  
>
Doesn't have to be. Remember, the scenario we're working from is someone 
faking a sender address in their otherwise normal endpoint, without the 
ability (or requirement) of intervening in normal network operations or 
altering DNS for the domain in question.


>Alternative call flow using the Identity mechanism falls into two cases, one
>where B trusts A with B's private key. This would be the case for voice mail
>systems, large gateways, conference servers, and such but not typically the
>case for phones other than in domains with a very small number of phones.
>Keep in mind this thread started around domains that were too cheap to get a
>certificate from a well known CA.
>
>  
>
Huh? I doubt Cisco is going to hand you the private key to their proxy 
so that you can use it to sign requests. Consequently we fall into the 
second case for all real deployments.

>If B trusts A, then A can insert the identity header, send the call to C. If
>the cert is from a well known CA then CA then C can verify the identity of
>the sender. Note number of messages involved was 1.
>
>If B does not trust A, then A can send call to B using previous digest
>credential and B can insert identity header and send call to C and C can
>verify it. Message count 2.
>
>If B does not have a cert from a well known CA, it can still do the same
>things as before but C will need to fetch the cert from B. The *first* time
>this happens we add a +2 to the message count but it can be cached for later
>messages. 
>
>  
>
Excellent point. If C trusts that B is authoritative for the domain of 
A, then C has as much reason to trust a cert delivered by B as it does 
to trust an assertion on A's behalf made by B. This obviates the need 
for dial backs (except, of course, for certificate revocation issues)  
for the life of the  cert, if we use an identity-assertion-by-signing 
approach.

Now, what does this get us that  just having C verify 1) That B is 
authoritative for the domain of A according to DNS, and that 2) that the 
request came from the IP address of B, according to DNS, to the extent 
that a TCP socket could from.

You're doing the same exact thing, when you have C trust DNS and the 
routing path to B in order to fetch the cert from B.

As far, as I know, all the crypto you've outlined falls back to 
assumptions that are just as supportable in the non-crypto solution I 
counter-proposed to Juha,  with a lot less CPU effort and network 
traffic required.

One different property is that both assertion-by-signing and 
assertion-by-dialback can be "checked" multiple hops down the 
router-chain, whereas assertion-by-source-address cannot. Is this 
critical for real-world deployments?

... section elided ...

>  
>
>>But the (relatively) important thing about this approach is that it does
>>block attacks where somebody pretends to be A, but doesn't really have
>>the wherewithal to either authenticate to B, or to subvert the mostly
>>unprotected the B-C link. This is excatly the weakness that the majority
>>of source-forged spammers use to deliver spam in today's world.
>>
>>    
>>
>
>
>It's not clear to me what problem dialback solves that many deployments
>would care about. Today the B-C links are very well protected as they mostly
>run in IPSec tunnels between preconfigured links. The ones that are not all
>preconfigured can use mutual TLS.
>
>  
>
Well, last week I tried to send an email to somebody at RoadRunner, who 
bounced the email because the netblock (PTR) delegation for the address 
of my mail server du jour wasn't right. It turns out it's REALLY 
difficult for ISPs to get this right. Certainly the B-C link between my 
email server and their email server isn't going to be IPSec protected 
anytime soon. But if they'd been able to just look at DNS and say 
"Whaddya know. That email DID come through the MX for softarmor.com", 
things would have been ok. We don't do that for mail, because many 
companies run diassgregated inbound and outbound mail servers, and MX 
records only describe inbound servers.

And once again, remember that this thread was started by somebody (NOT 
ME ;-) ) who runs a REAL SIP NETWORK, is impacted by this problem in the 
real world,  is looking for an adequate, deployable solution, and hasn't 
been able to make registered certs appear to be financially attractive.

--
Dean

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From sip-bounces@ietf.org  Mon Oct 11 19:33:56 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
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Again, Dean, my immediate concern here is whether or not this discussion
about a hypothetical, partially-formulated alternative is material to the
progression of sip-identity-03.

While it is interesting to discuss alternatives that provide a weaker
assurance, I see no reason why the exploration of those alternatives should
impede the standardization of a system that seems to satisfy the broader
requirement.

Ultimately, this thread was kicked off with the contention that acquiring
certificates is infeasible. Both Cullen and I provided some data about
certificate acquisition that seemed to argue that the problem isn't nearly
so bad. If you can get a publicly-verifiable certificate for under $100 a
year, then the expenditure is on the same order of magnitude as acquiring a
domain name. I haven't yet heard the parallel argument that domain names are
so expensive and administratively cumbersome that we should invent a new
alternative system for locating SIP servers, but in my opinion, this
contention would carry roughly the same weight as the arguments I've heard
against certificates so far.

If there is in fact no problem with making certificates a component of the
identity architecture, then I'm not sure why we want to explore the
specification of alternatives. Alternatives are bad, when it comes to
security. I continue to believe that we should set a long-term goal of
arriving a single mandatory-to-use identity mechanism for SIP. Partial
solutions will not get us where we need to be.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, October 11, 2004 1:03 PM
> To: Peterson, Jon
> Cc: Cullen Jennings; sip@ietf.org; Juha Heinanen
> Subject: Re: [Sip] verifying identity of sender
> 
[snip]
> 
> So, does Juha's approach offer any security benefits over simpler hacks, 
> like my source-to-authority matching requirement? If it does, then let's 
> understand those properties and see if any sort of standardization is 
> needed. If not, then let's see what properties really are useful, and 
> see if we need to document practices that exploit those properties for 
> the operator community.
> 

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From sip-bounces@ietf.org  Mon Oct 11 20:24:23 2004
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Subject: [Sip] jon's rodeo of certificate savings!
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Let's see if this survives anyone's spam filters.

Here's a round-up of the certificate business as seen through Google this
afternoon. This is far from an exhaustive list - basically, it's as many as
I was able to go through before I got bored.

Enrollment price for a certificate (contracting lasting a single year) is
given - multi-year contracts generally provide a lower cost per annum. Only
the lowest-cost applicable certificates offered by a particular provider are
listed. I've tried to limit this to CAs whose root cert is currently
distributed with IE, but in some cases I couldn't be exactly sure (all
promise compatibility with over 95% of existing web browsers). Only
certificates that cost less than US$100 per year are listed. About half of
the CAs provide free 30-day trial certificates. 

This mail does not constitute an endorsement of any of these services.

RegisterFly - $25 (special offer: $9!)
http://registerfly.com/ssl/
("instant signup process (as little as 5 minutes)")

FreeSSL - $39 (special offer: $19 - act now, supplies are limited (by the
DNS namespace size)!)
http://www.freessl.com/
("issued in minutes, installed in seconds!")
Uses the StarterSSL root. Another reseller:
http://www.ev1servers.net/english/starterssldetails.asp
... is apparently selling StarterSSL certs for only $4.95!

GoDaddy SSL - $49 (special offer: $29)
http://www.godaddy.com/gdshop/ssl/turbo.asp?isc=sslgol003t
(certificate issued "within minutes")

InstantSSL - $49
http://www.instantssl.com/
("certificates can be issued instantly" thanks to some proprietary thingy
they run)

DigiCertSSL - $99
http://www.digicert.com/SSL-Server-certificatee.html

GeoTrust - $149 - but if you're currently a Thawte/Verisign customer, only
$69 to enroll!
http://www.geotrust.com/jump/upgrade-ssl-certificate.html
(promises to issue a certificate in 10 minutes or less, sort of like pizza I
guess)

Jon Peterson
NeuStar, Inc.

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Subject: [Sip] Queries/Comments on Resource Priority-04
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Hi,

I've taken a look at the draft and have the following comments:

General comments:
- The fact that it isn't mandatory to mark all responses with RP headers means that one should at least make it a SHOULD(MUST?) to provide "resource prioritization" based on a dialog or session initiated with the RP header (possibly requiring Stateful proxy?). Otherwise it seems that being able to know if a node "supports Resource priority" (using OPTIONS or a Require hdr) may not be of use if you can't be assured the whole dialog/session will be assured of the "resource prioritization"? If it's not specified in this document then I guess it will need to be specified in some external policy document...(out of scope)


Section 3.3
- minor comment: It might be useful to label table something like 'Table 1'.
- In the second the paragraph the 420 response is incorrectly defined as "(Not Supported)" - when it should be "(Bad Extension)".
- In the same paragraph the explanation following the handling of the 420 response seems at odds with the description in section 4.2.2 and that of RFC3261. Firstly this is because it indicates that a 420 is generated when it encounters a unknown RP value (on an RP capable node) -  shouldn't that be covered in section 4.2.2 - Shouldn't a 420 be generated only by non RP capable node in response to Require? Secondly it seems to reccomend adding an 'Accept-Resource-Priority' header when RFC3261 specifies adding an "Unsupported" header for a 420 response? Or maybe it was intended to talk about the 417 handling here?

Example 7.2
- In this example - according to section 4.2.2 a 417 response should only generated when in strict mode, in which case the shouldn't the INVITE contain the Require header?


Piers O'Hanlon
--------------_____________
Computer Science Department
University College London









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From sip-bounces@ietf.org  Tue Oct 12 14:22:13 2004
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Sorry to bother this list, but I couldn't find a clear statement
in the RFC or a consensus on the sip-implementors list.

For the UA that sends a BYE, is the dialog terminated when the
BYE is sent or when the 200 response is received?  Can you point
to a quote in the RFC?

Related questions:

Should crossed BYEs generate 481s or 200s?  (I think the answer is
obvious based on the answer to the first question.)

If the dialog is not terminated until the BYE's response is received,
how are re-INVITES handled during the period from when the BYE is sent
until the response is received?  Specifically the RFC refers to the
session/media being stopped when the BYE is sent.  Could a re-INVITE
restart the session/media?

Thanks
-troy

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Subject: Re: [Sip] precisely when is a dialog terminated
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Troy,
This topic has been hashed out a number of times. Refer to archives for discussion on this topic.

-suheel

-----------------
Suheel Hussain

----- Original Message -----
From: Troy Cauble <troy@bell-labs.com>
Date: Tuesday, October 12, 2004 1:09 pm
Subject: [Sip] precisely when is a dialog terminated

> 
> Sorry to bother this list, but I couldn't find a clear statement
> in the RFC or a consensus on the sip-implementors list.
> 
> For the UA that sends a BYE, is the dialog terminated when the
> BYE is sent or when the 200 response is received?  Can you point
> to a quote in the RFC?
> 
> Related questions:
> 
> Should crossed BYEs generate 481s or 200s?  (I think the answer is
> obvious based on the answer to the first question.)
> 
> If the dialog is not terminated until the BYE's response is received,
> how are re-INVITES handled during the period from when the BYE is sent
> until the response is received?  Specifically the RFC refers to the
> session/media being stopped when the BYE is sent.  Could a re-INVITE
> restart the session/media?
> 
> Thanks
> -troy
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Use sipping@ietf.org for new developments on the application of sip
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From sip-bounces@ietf.org  Tue Oct 12 19:35:19 2004
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I'd love to.  Got a searchable archive?
I couldn't find one off the links below.

http://www1.ietf.org/mail-archive/web/sip/current/index.html
https://www1.ietf.org/mailman//listinfo/sip

I also tried this google search with no relevant results:
	site: ietf.org BYE dialog sip-mailing-list

-troy




suheelh@nc.rr.com wrote:

> Troy,
> This topic has been hashed out a number of times. Refer to archives for discussion on this topic.
> 
> -suheel
> 
> -----------------
> Suheel Hussain
> 
> ----- Original Message -----
> From: Troy Cauble <troy@bell-labs.com>
> Date: Tuesday, October 12, 2004 1:09 pm
> Subject: [Sip] precisely when is a dialog terminated
> 
> 
>>Sorry to bother this list, but I couldn't find a clear statement
>>in the RFC or a consensus on the sip-implementors list.
>>
>>For the UA that sends a BYE, is the dialog terminated when the
>>BYE is sent or when the 200 response is received?  Can you point
>>to a quote in the RFC?
>>
>>Related questions:
>>
>>Should crossed BYEs generate 481s or 200s?  (I think the answer is
>>obvious based on the answer to the first question.)
>>
>>If the dialog is not terminated until the BYE's response is received,
>>how are re-INVITES handled during the period from when the BYE is sent
>>until the response is received?  Specifically the RFC refers to the
>>session/media being stopped when the BYE is sent.  Could a re-INVITE
>>restart the session/media?
>>
>>Thanks
>>-troy
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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To: "'Troy Cauble'" <troy@bell-labs.com>, <suheelh@nc.rr.com>
Subject: RE: [Sip] precisely when is a dialog terminated
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https://www1.ietf.org/mailman/listinfo/sip

guides you to:

http://www1.ietf.org/pipermail/sip/

Which you can plug into google (I think the trick is site:www1.ietf.org)

http://tinyurl.com/3ltlo

More or less works I think

-Tom

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Troy Cauble
> Sent: Tuesday, October 12, 2004 4:19 PM
> To: suheelh@nc.rr.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] precisely when is a dialog terminated
> 
> 
> I'd love to.  Got a searchable archive?
> I couldn't find one off the links below.
> 
> http://www1.ietf.org/mail-archive/web/sip/current/index.html
> https://www1.ietf.org/mailman//listinfo/sip
> 
> I also tried this google search with no relevant results:
> 	site: ietf.org BYE dialog sip-mailing-list
> 
> -troy
> 
> 
> 
> 
> suheelh@nc.rr.com wrote:
> 
> > Troy,
> > This topic has been hashed out a number of times. Refer to 
> archives for discussion on this topic.
> > 
> > -suheel
> > 
> > -----------------
> > Suheel Hussain
> > 
> > ----- Original Message -----
> > From: Troy Cauble <troy@bell-labs.com>
> > Date: Tuesday, October 12, 2004 1:09 pm
> > Subject: [Sip] precisely when is a dialog terminated
> > 
> > 
> >>Sorry to bother this list, but I couldn't find a clear statement in 
> >>the RFC or a consensus on the sip-implementors list.
> >>
> >>For the UA that sends a BYE, is the dialog terminated when 
> the BYE is 
> >>sent or when the 200 response is received?  Can you point 
> to a quote 
> >>in the RFC?
> >>
> >>Related questions:
> >>
> >>Should crossed BYEs generate 481s or 200s?  (I think the answer is 
> >>obvious based on the answer to the first question.)
> >>
> >>If the dialog is not terminated until the BYE's response is 
> received, 
> >>how are re-INVITES handled during the period from when the 
> BYE is sent 
> >>until the response is received?  Specifically the RFC refers to the 
> >>session/media being stopped when the BYE is sent.  Could a 
> re-INVITE 
> >>restart the session/media?
> >>
> >>Thanks
> >>-troy
> >>
> >>_______________________________________________
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP Protocol Use 
> >>sip-implementors@cs.columbia.edu for questions on current sip Use 
> >>sipping@ietf.org for new developments on the application of sip
> >>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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From sip-bounces@ietf.org  Wed Oct 13 15:12:24 2004
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From: Troy Cauble <troy@bell-labs.com>
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Tom's google tip returned a lot more results but
I didn't find any that addressed my questions.

I've read the RFC and the archives.
Anyone want to help out?  The primary question was:

 > For the UA that sends a BYE, is the dialog terminated when the
 > BYE is sent or when the 200 response is received?  Can you point
 > to a quote in the RFC?


The following quote from 15.1.1 says that the Session is terminated
when the BYE transaction is started, but doesn't clearly say when the
dialog is terminated.  The last sentence may imply that it's terminated
when a final response is received, but that's inferring a lot, IMO.

>    The UAC MUST
>    consider the session terminated (and therefore stop sending or
>    listening for media) as soon as the BYE request is passed to the
>    client transaction.  If the response for the BYE is a 481
>    (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no
>    response at all is received for the BYE (that is, a timeout is
>    returned by the client transaction), the UAC MUST consider the
>    session and the dialog terminated.


"Terminating a dialog" by "sending a BYE" is referred to more than
once, but I take that as specifying the mechanism, not the precise
timing.

Thanks,
-troy



Troy Cauble wrote:
> 
> Sorry to bother this list, but I couldn't find a clear statement
> in the RFC or a consensus on the sip-implementors list.
> 
> For the UA that sends a BYE, is the dialog terminated when the
> BYE is sent or when the 200 response is received?  Can you point
> to a quote in the RFC?
> 
> Related questions:
> 
> Should crossed BYEs generate 481s or 200s?  (I think the answer is
> obvious based on the answer to the first question.)
> 
> If the dialog is not terminated until the BYE's response is received,
> how are re-INVITES handled during the period from when the BYE is sent
> until the response is received?  Specifically the RFC refers to the
> session/media being stopped when the BYE is sent.  Could a re-INVITE
> restart the session/media?
> 
> Thanks
> -troy

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Hi All,=20

=20

      Is OPTIONS method should be part of the dialog, or can be before
initiation of dialog. From RFC, I understand that OPTIONS can be sent in
two scenarios. Please tell that it is correct.

=20

Thanks in Advance,

Anil

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Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

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From: "Shekar, Nagesh Soma (Shekar)** CTR **" <shek@lucent.com>
To: "'Anil Bollineni'" <ABollineni@juniper.net>, sip@ietf.org
Subject: RE: [Sip] [Sip-implementors] Question on OPTIONS method
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Anil,

OPTIONS transaction can be part of a exisiting Dialog or can 
create a new Dialog.

Nagesh Shekar

> -----Original Message-----
> From: Anil Bollineni [mailto:ABollineni@juniper.net]
> Sent: Wednesday, October 13, 2004 3:25 PM
> To: sip@ietf.org
> Subject: [Sip] [Sip-implementors] Question on OPTIONS method
> 
> 
> 
> Hi All, 
> 
>  
> 
>       Is OPTIONS method should be part of the dialog, or can be before
> initiation of dialog. From RFC, I understand that OPTIONS can 
> be sent in
> two scenarios. Please tell that it is correct.
> 
>  
> 
> Thanks in Advance,
> 
> Anil
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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Thanks, I got the same opinion form RFC.

Anil

-----Original Message-----
From: Shekar, Nagesh Soma (Shekar)** CTR ** [mailto:shek@lucent.com]=20
Sent: Wednesday, October 13, 2004 12:50 PM
To: Anil Bollineni; sip@ietf.org
Subject: RE: [Sip] [Sip-implementors] Question on OPTIONS method

Anil,

OPTIONS transaction can be part of a exisiting Dialog or can=20
create a new Dialog.

Nagesh Shekar

> -----Original Message-----
> From: Anil Bollineni [mailto:ABollineni@juniper.net]
> Sent: Wednesday, October 13, 2004 3:25 PM
> To: sip@ietf.org
> Subject: [Sip] [Sip-implementors] Question on OPTIONS method
>=20
>=20
>=20
> Hi All,=20
>=20
> =20
>=20
>       Is OPTIONS method should be part of the dialog, or can be before
> initiation of dialog. From RFC, I understand that OPTIONS can=20
> be sent in
> two scenarios. Please tell that it is correct.
>=20
> =20
>=20
> Thanks in Advance,
>=20
> Anil
>=20
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From sip-bounces@ietf.org  Wed Oct 13 18:23:36 2004
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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To: Troy Cauble <troy@bell-labs.com>
Subject: Re: [Sip] precisely when is a dialog terminated
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Troy Cauble wrote:

> The following quote from 15.1.1 says that the Session is terminated
> when the BYE transaction is started, but doesn't clearly say when the
> dialog is terminated.  The last sentence may imply that it's terminated
> when a final response is received, but that's inferring a lot, IMO.

Troy:

Given that the SIP community believes that dialog sharing
may be undesirable, you can probably rest assured that the
dialog is terminated when a final response is received.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From sip-bounces@ietf.org  Thu Oct 14 10:23:47 2004
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] precisely when is a dialog terminated
Date: Thu, 14 Oct 2004 09:19:34 -0500
To: "Vijay K. Gurbani" <vkg@lucent.com>
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What is the practical impact of the answer to this question on your 
application?

For BYE, the dialog is going to terminate (for a given endpoint) when a 
200 is
sent or received. Why? You could get a recoverable error (a 407 or a 
503 for
example) response from an intervening proxy, and you need the dialog 
state
to generate your next request.

You can use similar reasoning to determine when you can lose state for 
other
dialog terminating events.

RjS

On Oct 13, 2004, at 4:45 PM, Vijay K. Gurbani wrote:

> Troy Cauble wrote:
>
>> The following quote from 15.1.1 says that the Session is terminated
>> when the BYE transaction is started, but doesn't clearly say when the
>> dialog is terminated.  The last sentence may imply that it's 
>> terminated
>> when a final response is received, but that's inferring a lot, IMO.
>
> Troy:
>
> Given that the SIP community believes that dialog sharing
> may be undesirable, you can probably rest assured that the
> dialog is terminated when a final response is received.
>
> - vijay
> -- 
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Wireless Networks Group/Internet Software and Services
> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224 0216
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org  Fri Oct 15 06:33:19 2004
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SUBJECT: Re: Re: [Sip] Problem with an intial SIP INVITE server transa ction.
DATE: 27 Sep 2004 15:50:40 GMT
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I do not belive this will work. In case of proxy it can route this INVIT=
E message to a completely different endpoint, then it did for the first =
INVITE. This means that a new dialog will be established, which is clear=
ly not intended=0D=0A=0D=0AAs far as SIP UA is concerned, this will also=
 not work, for one very simple reason -- SIP INVITE does not match the S=
IP dialog. If we want to create a mechanism to match this INVITE to an e=
xisting dialog, this mechanism should be part of the standard. Furthermo=
re, it would be a very bad thing to create a new transaction for the re-=
transmitted INVITE. Imagine, what would the SIP UA, that originated the =
call do if it receives two responses, one successful and one failure in =
unspecified order. If SIP UA will receive the failure or out of order re=
sponse first, it will ignore all the success responses afterwards. I am =
not even sure what SIP UA will do if it receives a failure response afte=
r the success. I do not belive this is even allowed.=0D=0A=0D=0AAs I men=
tioned before, the most consistent way of dealing with this scenario is =
to put a timer, which will keep the transaction going after 2XX response=
 is sent. If transaction receives a message during this time, this messa=
ge is ignored. =0D=0A=0D=0AAs I mentioned before, I am not looking as mu=
ch for the way to deal with this problem, as for the way to fix the RFC.=
=0D=0A___________________________________=0D=0ARoman Shpount, VP of Tech=
nology=0D=0AaTelo, Inc. -- www.atelo.com=0D=0A=0D=0A------ Original mess=
age ------=0D=0A=0D=0A>From: Gautham <gautham@mistralsoftware.com>=0D=0A=
>Subject: Re: [Sip] Problem with an intial SIP INVITE server transa ctio=
n.=0D=0A>Date: 27 Sep 04, 02:18 PM=0D=0A>To: Paul D.Smith <Paul.D.Smith@=
dataconnection.com>=0D=0A>Cc: 'Roman Shpount' <roman@atelo.com>, SIP WG =
<sip@ietf.org>=0D=0A=0D=0AHi,=0D=0A=0D=0AThe scenario given by Roman cou=
ld happen at either a proxy or a endpoint.=0D=0A=0D=0AAt a proxy, the ap=
propriate thing would be to create a new transaction.=0D=0ALike Roman ha=
s mentioned in his mail, this retransmission may be because =0D=0Athe=0D=
=0A1xx responses and the 2xx responses were lost. In such a case,  I bel=
ieve,=0D=0Ait would be correct for the proxy to create a new transaction=
 and=0D=0Aforward the INVITE downstream, as if it were a new transaction=
.=0D=0AIn anycase,  a proxy would not know better.=0D=0A=0D=0AAt an endp=
oint too, it should not be a problem if a new server =0D=0Atransaction g=
ets created=0D=0Afor the retransmitted INVITE. The UA core can look at t=
he From  tag, =0D=0ACall-Id and possibly CSeq=0D=0Ato determine that it =
is not a new dialog request i.e., that it matches =0D=0Aan ongoing dialo=
g.=0D=0AIf the UA core is still retransmitting the 2xx, it would send th=
e 2xx to =0D=0Athe=0D=0Atransaction which would terminate it. If the UA =
core has finished =0D=0Aretransmitting=0D=0Athe 2xx (and has received th=
e ACK) it could treat the INVITE as an=0D=0Aout-of-sequence request.=0D=
=0A=0D=0AHope this helps.=0D=0A=0D=0ARegards,=0D=0AA. N. Gautham=0D=0A=
=0D=0APaul D.Smith wrote:=0D=0A=0D=0A>Roman,=0D=0A>=0D=0A>I believe you =
have strayed into a poorly specified area of SIP here.  I have=0D=0A>als=
o found nothing in the specs. that directly answers your question howeve=
r=0D=0A>I can tell you what is observed:  the duplication INVITEs are si=
mply=0D=0A>ignored.  One option is to keep track of the INVITE transacti=
on until the=0D=0A>ACK arrives.  Although logically there is no transact=
ion layer interaction=0D=0A>(as per section 17, RFC 3261), you can then =
identify the INVITE as a=0D=0A>duplicate and discard it.=0D=0A>=0D=0A>Pa=
ul D Smith=0D=0A>Network Protocols Group =0D=0A>Data Connection Ltd (DCL=
)=0D=0A>Tel: +44 20 8366 1177  Email: paul.d.smith@dataconnection.com =
=0D=0A>Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com=0D=0A=
>=0D=0A>=0D=0A>-----Original Message-----=0D=0A>From: Roman Shpount [mai=
lto:roman@atelo.com]=0D=0A>Sent: 19 September 2004 08:15=0D=0A>To: SIP P=
ublic Folder (E-mail)=0D=0A>Subject: Re: RE: RE: [Sip] Problem with an i=
ntial SIP INVITE server=0D=0A>transaction.=0D=0A>=0D=0A>=0D=0A>Thomas,=
=0D=0A>=0D=0A>The problem is that retransmitted INVITE message does not =
have the To-tag.=0D=0A>Thus it does not match the dialog and UAC is not =
even informed about this=0D=0A>message. The only way I could think to de=
al with this, as I described in my=0D=0A>original email, was to add a ti=
mer which keeps the server INVITE transaction=0D=0A>open for T4 after 2X=
X response is sent. If retransmitted INVITE message=0D=0A>arrives within=
 this time, it matches the transaction and is ignored. By the=0D=0A>way,=
 this discussion is not academic, this is the real problem that we=0D=0A=
>encountered while dealing with large volumes of SIP transactions over W=
i-Fi.=0D=0A>___________________________________=0D=0A>Roman Shpount, VP =
of Technology=0D=0A>aTelo, Inc. -- www.atelo.com=0D=0A>=0D=0A>------ Ori=
ginal message ------=0D=0A>=0D=0A>  =0D=0A>=0D=0A>>From: Thomas Gal <Tho=
masGal@LumenVox.com>=0D=0A>>Subject: RE: RE: [Sip] Problem with an intia=
l SIP INVITE server=0D=0A>>    =0D=0A>>=0D=0A>transaction.=0D=0A>  =0D=
=0A>=0D=0A>>Date: 31 Aug 04, 04:44 PM=0D=0A>>To: 'Roman Shpount' <roman@=
atelo.com>,  <gledgard@iperia.com>,=0D=0A>>    =0D=0A>>=0D=0A><sip@ietf.=
org>=0D=0A>=0D=0A>Maybe some people were just trying to help without sco=
uring the spec for all=0D=0A>the details. I have read the spec quite a f=
ew times, thanks, but here, maybe=0D=0A>this will help(17):=0D=0A>=0D=0A=
>	In the case of a transaction where the=0D=0A>   request was an INVITE =
(known as an INVITE transaction), the=0D=0A>   transaction also includes=
 the ACK only if the final response was not=0D=0A>   a 2xx response.  If=
 the response was a 2xx, the ACK is not considered=0D=0A>   part of the =
transaction.=0D=0A>=0D=0A>      The reason for this separation is rooted=
 in the importance of=0D=0A>      delivering all 200 (OK) responses to a=
n INVITE to the UAC.  To=0D=0A>      deliver them all to the UAC, the UA=
S alone takes responsibility=0D=0A>      for retransmitting them (see Se=
ction 13.3.1.4), and the UAC alone=0D=0A>      takes responsibility for =
acknowledging them with ACK (see Section=0D=0A>      13.2.2.4).  Since t=
his ACK is retransmitted only by the UAC, it is=0D=0A>      effectively =
considered its own transaction.=0D=0A>=0D=0A>You gotta wait til you get =
the ACK.=0D=0A>=0D=0A>-Tom=0D=0A>=0D=0A>=0D=0A>-----Original Message----=
-=0D=0A>From: Roman Shpount [mailto:roman@atelo.com] =0D=0A>Sent: Tuesda=
y, August 31, 2004 8:38 AM=0D=0A>To: gledgard@iperia.com; ThomasGal@Lume=
nVox.com; sip@ietf.org=0D=0A>Subject: Re: RE: [Sip] Problem with an inti=
al SIP INVITE server transaction.=0D=0A>=0D=0A>Hi All, =0D=0A>=0D=0A>I g=
ot quite a few answers like: "there is a magic key which prevents this"=
=0D=0A>or "there is a timer that prevents this", or "you cannot get an I=
NVITE once=0D=0A>you sent the 1XX or 2XX response". I would suggest that=
 you would read the=0D=0A>RFC (Section 17.2.1) before rushing to answer =
my question. RFC clearly=0D=0A>spells out that INVITE server transaction=
 is terminated when 2XX response is=0D=0A>sent. There is even a picture =
(Figure 7) that shows just that. There are no=0D=0A>timers. There are no=
 special transaction IDs. Transaction is ended, clear=0D=0A>and simple. =
Once transaction is ended, re-transmitted INVITE message should=0D=0A>st=
art the new transaction.  =0D=0A>___________________________________=0D=
=0A>Roman Shpount, VP of Technology=0D=0A>aTelo, Inc. -- www.atelo.com=
=0D=0A>=0D=0A>------ Original message ------=0D=0A>=0D=0A>  =0D=0A>=0D=
=0A>>From: Gordon Ledgard <gledgard@iperia.com>=0D=0A>>Subject: RE: [Sip=
] Problem with an intial SIP INVITE server transaction.=0D=0A>>Date: 31 =
Aug 04, 12:40 PM=0D=0A>>To:  <ThomasGal@LumenVox.com>, Roman Shpount <ro=
man@atelo.com>,  =0D=0A>><sip@ietf.org>=0D=0A>>    =0D=0A>>=0D=0A>=0D=0A=
>=0D=0A>Read the stuff about timers in the transaction layer. It spells =
out how you=0D=0A>need to hold on to transactions for messages over unre=
liable transports for=0D=0A>just this sort of reason.=0D=0A>=0D=0A>=0D=
=0A>Gordon=0D=0A>=0D=0A>=0D=0A>=0D=0A>-----Original Message-----=0D=0A>F=
rom: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On Behalf Of Thom=
as=0D=0A>Gal=0D=0A>Sent: Monday, August 30, 2004 7:54 PM=0D=0A>To: 'Roma=
n Shpount'; sip@ietf.org=0D=0A>Subject: RE: [Sip] Problem with an intial=
 SIP INVITE server transaction.=0D=0A>=0D=0A>=0D=0A>I'm not an expert on=
 this protocol (only read through the spec a few times=0D=0A>so it blurs=
 together with some of the other ones) but even without looking=0D=0A>I'=
m SURE there's supposed to be a unique request-ID generated by the clien=
t=0D=0A>to avoid just such a confusion. Of course in this case any state=
ful UA would=0D=0A>ignore the already processed invite request. Also (su=
re about this one) if=0D=0A>the client received the provisional(100) res=
ponse it wouldn't re-transmit=0D=0A>the INVITE either as that's the whol=
e point.=0D=0A>=0D=0A>-Tom=0D=0A>=0D=0A>-----Original Message-----=0D=0A=
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of R=
oman=0D=0A>Shpount=0D=0A>Sent: Tuesday, August 17, 2004 7:05 PM=0D=0A>To=
: sip@ietf.org=0D=0A>Subject: [Sip] Problem with an intial SIP INVITE se=
rver transaction.=0D=0A>=0D=0A>I run into the problem with handling of r=
etransmitted SIP INVITE message=0D=0A>based on the RFC 3621. =0D=0A>=0D=
=0A>Imagine the following situation=0D=0A>=0D=0A>INVITE 1=0D=0A>--------=
--------------->=0D=0A>100=0D=0A><-----------------------=0D=0A>2XX=0D=
=0A><-----------------------=0D=0A>re-transmitted INVITE 1=0D=0A>-------=
---------------->=0D=0A>=0D=0A>Based on RFC 3261, server INVITE transact=
ion terminates as soon as 2XX=0D=0A>response is sent. This means that re=
-transmitted INVITE will be treated as a=0D=0A>new transaction. This INV=
ITE message will not match the existing dialog,=0D=0A>since its To tag i=
s empty. This means it will be treated as new dialog=0D=0A>creating mess=
age and phone will treat this message as a new call, which is=0D=0A>clea=
rly not intended.=0D=0A>=0D=0A>This situation can occur if both 100 and =
2XX responses were lost by the=0D=0A>unreliable transport, and INVITE co=
ntinues to be re-transmitted. The fact=0D=0A>that, 100 and 2XX responses=
 were lost is unknown to the server transaction=0D=0A>and INVITE re-tran=
smit can arrive before the 2XX re-transmit is sent. =0D=0A>=0D=0A>Majori=
ty of the transaction state machines in SIP are designed to handle=0D=0A=
>messages, that are received after the response is sent. It seems that=
=0D=0A>something simular is required in this case as well. =0D=0A>=0D=0A=
>In our implementation of SIP stack, server transaction is not terminate=
d=0D=0A>when the 2XX response is sent. Instead, transaction is moved int=
o the=0D=0A>"ConfirmedBis" state and timer, which is set to fire in T4, =
is started. If=0D=0A>the stack receives an INVITE message which matches =
the server transaction in=0D=0A>this state, it ignores it. If stack rece=
ives a 2XX response, which matches=0D=0A>the server transaction, it pass=
es it to TU. Once the T4 timer is fired, the=0D=0A>transaction is termin=
ated.=0D=0A>=0D=0A>There is also an additional issue related to this pro=
blem. It looks like,=0D=0A>for the sake of consistency, some type of res=
ponse should be sent=0D=0A>immediately to the transport layer when INVIT=
E re-transmit is received. If=0D=0A>multiple 2XX responses has been sent=
, it is unclear what the behavior should=0D=0A>be: should all the 2XX me=
ssages be sent, should the last 2XX message be=0D=0A>resent or should th=
e INVITE be plainly ignored. =0D=0A>=0D=0A>_____________________________=
______=0D=0A>Roman Shpount, VP of Technology=0D=0A>aTelo, Inc. -- www.at=
elo.com=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>________________=
_______________________________=0D=0A>Sip mailing list  https://www1.iet=
f.org/mailman/listinfo/sip=0D=0A>This list is for NEW development of the=
 core SIP Protocol Use=0D=0A>sip-implementors@cs.columbia.edu for questi=
ons on current sip Use=0D=0A>sipping@ietf.org for new developments on th=
e application of sip=0D=0A>=0D=0A>=0D=0A>_______________________________=
________________=0D=0A>Sip mailing list  https://www1.ietf.org/mailman/l=
istinfo/sip=0D=0A>This list is for NEW development of the core SIP Proto=
col Use=0D=0A>sip-implementors@cs.columbia.edu for questions on current =
sip Use=0D=0A>sipping@ietf.org for new developments on the application o=
f sip=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>=0D=
=0A>=0D=0A>=0D=0A>=0D=0A>=0D=0A>________________________________________=
_______=0D=0A>Sip mailing list  https://www1.ietf.org/mailman/listinfo/s=
ip=0D=0A>This list is for NEW development of the core SIP Protocol=0D=0A=
>Use sip-implementors@cs.columbia.edu for questions on current sip=0D=0A=
>Use sipping@ietf.org for new developments on the application of sip=0D=
=0A>=0D=0A>_______________________________________________=0D=0A>Sip mai=
ling list  https://www1.ietf.org/mailman/listinfo/sip=0D=0A>This list is=
 for NEW development of the core SIP Protocol=0D=0A>Use sip-implementors=
@cs.columbia.edu for questions on current sip=0D=0A>Use sipping@ietf.org=
 for new developments on the application of sip=0D=0A>=0D=0A>  =0D=0A>=
=0D=0A=0D=0A=0D=0A=0D=0A=0D=0A=0D=0A=0D=0A


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From sip-bounces@ietf.org  Fri Oct 15 06:34:35 2004
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Content-class: urn:content-classes:message
Subject: RE: [Sip] Symmetric NAT issue
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Thread-Topic: [Sip] Symmetric NAT issue
thread-index: AcSnbWzg4kYiaxdgQkahlkF5HEkMigAE0YcQ
From: "Christian Stredicke" <Christian.Stredicke@snom.de>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]=20
>=20
> Christian Stredicke wrote:
>=20
> > IMVHO SBC that use ICE are a smart alternative to using TURN.=20
>=20
> I don't really understand what this means. I would not expect=20
> an SBC to=20
> implement ICE, in the sense that it would do the p2p STUN pings,=20
> connectivity checks, etc., or are you proposing that it does?

In ICE we are talking about possible communication addresses, right? One
possible address is the private address, another adress that has been
allocated via a STUN packet sent out to a STUN server. The trick is that
AFTER the UA has created the SDP with these possible addresses and sent
them to the SBC, the SBC ADDS another communication address:

UA sends SDP:

v=3D0
o=3D17187614121 8000 8000 IN IP4 <private ip address>=20
s=3DSIP Call=20
c=3DIN IP4 <private ip address>=20
t=3D0 0=20
m=3Daudio 49242 RTP/AVP 0=20
a=3Drtpmap:0 PCMU/8000
a=3Dalt: <private ip address>
a=3Dalt: <stun ip address>

The SBC allocates RTP relay ports and MODIFIES the SDP:

v=3D0
o=3D17187614121 8000 8000 IN IP4 <SBC ip address>=20
s=3DSIP Call=20
c=3DIN IP4 <SBC ip address>=20
t=3D0 0=20
m=3Daudio 49242 RTP/AVP 0=20
a=3Drtpmap:0 PCMU/8000
a=3Dalt: <private ip address>
a=3Dalt: <stun ip address>
a=3Dalt: <SBC ip address>

The STUN packets and everything else is just forwarded, the SBC does not
implemenent any logic for that. That means when ICE tries the
connections, it will give a realistic picture of the delay. If the RTP
has not started yet and the SBC could not get the IP/port of the
symmetrical NAT, it will not relay that packet. But thats no problem as
the primary contact is the SBC allocated port (c-line).

>=20
> Also, ICE makes use of TURN, to deal with the symmetric NAT=20
> case. SO I=20
> don't understand what you mean when you say ICE is an alternative to=20
> TURN. Can you explain?
>=20
Tried to explain above.
>=20
> UA that
> > are not aware of ICE, STUN, TURN or NAT will run media=20
> through the B2BUA
> > while ICE-aware devices can bypass them.
>=20
> That I agree with.
>=20
> > IMEHO ist much easier to
> > implement STUN/ICE than TURN (well, we did/tried it) and it=20
> solves the
> > problem with the "dumb" devices.
>=20
> ICE without TURN will not help in the symmetric NAT case.

See above.

>=20
>=20
> > Paired with DNS SRV and STUN its
> > possible to automatically detect the nearest SBC, in case the SBC is
> > necessary to relay media for symmetrical NAT.
>=20
> Detect an SBC? I don't follow.

Assume that you are a global operator with customers in USA, Australia,
Europe and China. Customers in China expect that the delay when talking
in China is reasonable (no relaying via USA). That means the UA must use
the China SBC for communication. You can detect that by probing the
possible SBC with STUN packets, the addresses are listed in DNS SRV. You
just send a packet to all of them and pick the one with the shortest
delay.

This is necessary because you cannot for the INVITE to all DNS SRV
listed SBC. Here TURN would be better, cause you can allocate ports on
all TURN servers and ICE will then find the shortest delay.

Take a look at http://snom.com/download/natf_205.pdf. Henry forbid me to
use the three bad letters S-B-C...=20

Keep smiling, CS

>=20
> Thanks,
> Jonathan R.
>=20
>=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
>=20
>=20
>=20

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From sip-bounces@ietf.org  Fri Oct 15 11:06:35 2004
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Message-ID: <416FE437.7090804@bell-labs.com>
Date: Fri, 15 Oct 2004 10:52:39 -0400
From: Troy Cauble <troy@bell-labs.com>
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Subject: Re: [Sip] precisely when is a dialog terminated
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Thank you!  The possibility of a 407 clarifies things nicely.
-troy

Robert Sparks wrote:

> What is the practical impact of the answer to this question on your 
> application?
> 
> For BYE, the dialog is going to terminate (for a given endpoint) when a 
> 200 is
> sent or received. Why? You could get a recoverable error (a 407 or a 503 
> for
> example) response from an intervening proxy, and you need the dialog state
> to generate your next request.
> 
> You can use similar reasoning to determine when you can lose state for 
> other
> dialog terminating events.
> 
> RjS
> 
> On Oct 13, 2004, at 4:45 PM, Vijay K. Gurbani wrote:
> 
>> Troy Cauble wrote:
>>
>>> The following quote from 15.1.1 says that the Session is terminated
>>> when the BYE transaction is started, but doesn't clearly say when the
>>> dialog is terminated.  The last sentence may imply that it's terminated
>>> when a final response is received, but that's inferring a lot, IMO.


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Isn't it also necessary to to ensure that there are no outstanding 
subscriptions
that  have application states associated with the dialog?

See the following from rfc 3265, section 3.3.4. Dialog creation and 
termination :

   If a subscription's destruction leaves no other application state
   associated with the dialog, the dialog terminates.  The destruction
   of other application state (such as that created by an INVITE) will
   not terminate the dialog if a subscription is still associated with
   that dialog.

      Note that the above behavior means that a dialog created with an
      INVITE does not necessarily terminate upon receipt of a BYE.
      Similarly, in the case that several subscriptions are associated
      with a single dialog, the dialog does not terminate until all the
      subscriptions in it are destroyed.


Cheers,
Alex.


Troy Cauble wrote:

> Thank you!  The possibility of a 407 clarifies things nicely.
> -troy
>
> Robert Sparks wrote:
>
>> What is the practical impact of the answer to this question on your 
>> application?
>>
>> For BYE, the dialog is going to terminate (for a given endpoint) when 
>> a 200 is
>> sent or received. Why? You could get a recoverable error (a 407 or a 
>> 503 for
>> example) response from an intervening proxy, and you need the dialog 
>> state
>> to generate your next request.
>>
>> You can use similar reasoning to determine when you can lose state 
>> for other
>> dialog terminating events.
>>
>> RjS
>>
>> On Oct 13, 2004, at 4:45 PM, Vijay K. Gurbani wrote:
>>
>>> Troy Cauble wrote:
>>>
>>>> The following quote from 15.1.1 says that the Session is terminated
>>>> when the BYE transaction is started, but doesn't clearly say when the
>>>> dialog is terminated.  The last sentence may imply that it's 
>>>> terminated
>>>> when a final response is received, but that's inferring a lot, IMO.
>>>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org  Sat Oct 16 18:02:59 2004
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] precisely when is a dialog terminated
Date: Sat, 16 Oct 2004 16:59:34 -0500
To: alex.audu@alcatel.com
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So, the original question was about a dialog with one usage.

In general, a dialog goes away when its last usage goes away.

Each type of usage gets terminated in different ways, but the
same logic can be used for determining when that is.

See the -nit- drafts for a more detailed discussion.

RjS

On Oct 15, 2004, at 3:42 PM, Alex Audu wrote:

> Isn't it also necessary to to ensure that there are no outstanding 
> subscriptions
> that  have application states associated with the dialog?
>
> See the following from rfc 3265, section 3.3.4. Dialog creation and 
> termination :
>
>   If a subscription's destruction leaves no other application state
>   associated with the dialog, the dialog terminates.  The destruction
>   of other application state (such as that created by an INVITE) will
>   not terminate the dialog if a subscription is still associated with
>   that dialog.
>
>      Note that the above behavior means that a dialog created with an
>      INVITE does not necessarily terminate upon receipt of a BYE.
>      Similarly, in the case that several subscriptions are associated
>      with a single dialog, the dialog does not terminate until all the
>      subscriptions in it are destroyed.
>
>
> Cheers,
> Alex.
>
>
> Troy Cauble wrote:
>
>> Thank you!  The possibility of a 407 clarifies things nicely.
>> -troy
>>
>> Robert Sparks wrote:
>>
>>> What is the practical impact of the answer to this question on your 
>>> application?
>>>
>>> For BYE, the dialog is going to terminate (for a given endpoint) 
>>> when a 200 is
>>> sent or received. Why? You could get a recoverable error (a 407 or a 
>>> 503 for
>>> example) response from an intervening proxy, and you need the dialog 
>>> state
>>> to generate your next request.
>>>
>>> You can use similar reasoning to determine when you can lose state 
>>> for other
>>> dialog terminating events.
>>>
>>> RjS
>>>
>>> On Oct 13, 2004, at 4:45 PM, Vijay K. Gurbani wrote:
>>>
>>>> Troy Cauble wrote:
>>>>
>>>>> The following quote from 15.1.1 says that the Session is terminated
>>>>> when the BYE transaction is started, but doesn't clearly say when 
>>>>> the
>>>>> dialog is terminated.  The last sentence may imply that it's 
>>>>> terminated
>>>>> when a final response is received, but that's inferring a lot, IMO.
>>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>


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From sip-bounces@ietf.org  Mon Oct 18 09:52:25 2004
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From: Ramalakshmi M <ramalakshmi@ncoretech.com>
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Subject: [Sip] DNS support for Sip 3261 stack
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Hi All,

I want to implement DNS support for SIP 3261stack.
Can you please provide me some details regarding this.
    Which RFCs i should refer ?
    Any open source implementations will support  DNS in SIP 3261 Stack?
    Any application available on the net will support DNS so that i can  
i see the log of those packets ?

Right now i am referring,
     RFC 2782 - A DNS RR for specifying the location of services (DNS SRV)
     RFC 3263 - Session Initiation Protocol (SIP): Locating SIP Servers
     RFC 2168 - Resolution of Uniform Resource Identifiers using the 
Domain Name System

Thanks in Advance.

Regards,
Ramalakshmi.

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I will be out of the office starting  10/18/2004 and will not return until
10/25/2004.

I will respond to your message when I return.

Regards,
Nabeel



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From sip-bounces@ietf.org  Tue Oct 19 22:32:39 2004
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From: Anurag Kabra <anuragk@aftek.com>
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Subject: [Sip] Call Transfer Scenario
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Hi

     I need to clear few of my doubts regarding call transfer,=20

In a scenario of A, B and C - Unattended Transfer (Blind Transfer)

A =3D transferor
B =3D transferee
C =3D transfer target

1. There is a call set up between A and B
2.  A need to transfer to C - A puts the RTP session on Hold with Re-INVI=
TE=20
and sends REFER with refer-to C,=20
3. B on getting refer needs to put audio streaming on hold and so sends =20
Re-INVITE!! -- (DOUBT - is this step right)
4. B sends NOTIFY with subscription-state =3D active and expires to A
5. B now INVITES C=20
6. On getting response its sends a NOTIFY to A with subscription-state =3D=
=20
terminated
(Need to know if the REFER expires because of expires in NOTIFY what can =
be=20
done ie if we get a time out on A side)
7. Now A sends a BYE to B
(What to do if B wants to terminate the dialog with CANCEL or BYE)

How can we differentiate Attended Transfer and Blind Transfer in the=20
application??

Can anyone tell me free sip UA implementing Call Transfer?

Please do let me know any of your views. Please correct me if i am wrong=20
anywhere.=20

Best Regards
  Anurag.

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From: Colin Perkins <csp@csperkins.org>
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Subject: [Sip] Review request: draft-ietf-mmusic-connection-precon-00.txt
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Folks,

The MMUSIC working group is considering a draft on 
Connection-Establishment Preconditions in SIP 
<draft-ietf-mmusic-connection-precon-00.txt>. As there are obvious 
implications for SIP, this note is to solicit review by the SIP working 
group before we advance this draft in MMUSIC.

We'd be grateful to receive any comments - both for or against this 
draft - in reply to this message.

Thanks,
Colin


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From sip-bounces@ietf.org  Wed Oct 20 00:41:19 2004
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Peterson, Jon wrote:

>Again, Dean, my immediate concern here is whether or not this discussion
>about a hypothetical, partially-formulated alternative is material to the
>progression of sip-identity-03.
>
>  
>
Ahah! No, this has NOTHING to do with impeding the Identity draft, as 
far as I know. But it is a suggestion of another mechanism that MIGHT be 
useful to talk about. As far as I know, it NEVER had anything to do with 
the identity draft, but was a suggestion of another technique that might 
also be useful, and deployable in the nearer term. I think we've roughly 
concluded that the specific idea suggested has little or no benefit over 
a simpler operatioal procedure, but the exercise of that analysis is 
probably something useful for the larger community.

>While it is interesting to discuss alternatives that provide a weaker
>assurance, I see no reason why the exploration of those alternatives should
>impede the standardization of a system that seems to satisfy the broader
>requirement.
>  
>

Me neither.

>Ultimately, this thread was kicked off with the contention that acquiring
>certificates is infeasible. Both Cullen and I provided some data about
>certificate acquisition that seemed to argue that the problem isn't nearly
>so bad. If you can get a publicly-verifiable certificate for under $100 a
>year, then the expenditure is on the same order of magnitude as acquiring a
>domain name. I haven't yet heard the parallel argument that domain names are
>so expensive and administratively cumbersome that we should invent a new
>alternative system for locating SIP servers, but in my opinion, this
>contention would carry roughly the same weight as the arguments I've heard
>against certificates so far.
>
>  
>
Yes, but this thread SUGGESTED operational policies that, in the absence 
of universal certificate deployment, might cut down on sip-spam.

>If there is in fact no problem with making certificates a component of the
>identity architecture, then I'm not sure why we want to explore the
>specification of alternatives. Alternatives are bad, when it comes to
>security. I continue to believe that we should set a long-term goal of
>arriving a single mandatory-to-use identity mechanism for SIP. Partial
>solutions will not get us where we need to be.
>
>  
>
Operational procedures can be put in place in the "near end", (the 
domain individual operators are operating and protecting), and 
certificates gain little utility until they happen to be widely deployed 
at the FAR end, over which said operator has little control. That being 
said, there

--
Dean

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From sip-bounces@ietf.org  Wed Oct 20 01:01:19 2004
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Wearing my chair hat and thinking about draft-ietf-sip-identity-03, 
which IS an approved working-group effort that WAS scheduled to be 
finished last April . . .

 From the minutes at IETF 60: "It was observed that few participants in 
the room displayed confidence in their understanding of the problem or 
the proposal, and that further study would be a Good Thing."

Since then, we have talked a lot about identity-related issues on the list.

We've hashed through some of the less-understood issues, including the 
tremendous similarity between the SIP spam problem and the SMTP spam 
problem, and Michael Thomas pointed us at the MASS work, which is 
astoundingly similar to sip-identity.

Juha contributed some ideas on techniques related to jabber-style 
dial-back auth that could be applied to get partial identity in the 
absence of a certificate deployment, which led to discussion of 
additional operational hacks that could be applied to help with the 
problem, but neither gave us a complete solution.

Jon and Cullen proved that real CS-signed certificates are readily 
available at a cost not significantly different than that of a domain 
registration, or slightly less than the price of a decent bottle of 
Scotch, for some definition of the word "decent".

So, where do we go now? Are we ready to take this document to a working 
group last call? If you think not, please explain what we need to know, 
understand, or discuss in order to make forward progress here.

Thanks,

--
Dean Willis
Co-Chair, SIP Working Group

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From sip-bounces@ietf.org  Wed Oct 20 01:20:25 2004
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Subject: [Sip] WGLC for draft-sparks-sip-nit-problems-01.txt
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"chair hat on"

At San Diego, we had consensus that we're ready to discuss finalization 
of the non-invite trasnactions problem statement and corrective actions 
drafts.

Consequently, I'd like to request a Working Group Last Call (WGLC) on 
draft-sparks-sip-nit-problems-01.txt, which as I understand we intend to 
publish eventually as an informational RFC which will set up the 
near-term (as in NOW) work of the "actions" draft as well as establish 
problem definitions for other work we hope to undertake in the future, 
or at least think people should know about.

WGLC for ths draft will commence immediately, and I expect to be able to 
complete this review by November 3.

As always, please send your comments to the author (rjs@xten.com) and 
copy the list if you don't mind.

Thanks,

--
Dean Willis
SIP WG co-chair

_______________________________________________
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From sip-bounces@ietf.org  Wed Oct 20 01:33:14 2004
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"chair hat on still"

At San Diego, we had consensus that we're ready to discuss finalization 
of the non-invite transactions problem statement and corrective actions 
drafts.

Consequently, I'd like to request a Working Group Last Call (WGLC) on 
draft-sparks-sip-nit-actions-02, which as I understand we intend to 
publish eventually as an proposed standard RFC that will change a few 
aspects of RFC 3261 and related NIT request types to prevent some of the 
problems outlined in the problem statement draft.

WGLC for ths draft will commence immediately, and I expect to be able to 
complete this review by November 3.

As always, please send your comments to the author (rjs@xten.com) and 
copy the list if you don't mind.

Thanks,

--
Dean Willis
SIP WG co-chair

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From sip-bounces@ietf.org  Wed Oct 20 11:42:21 2004
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bob,

i've defined the following object to incorporate your suggestion
for monitoring rsp status code statistics on a per method basis.

+    sipStatusCodeMethod OBJECT-TYPE
+        SYNTAX     SipMethodIdentifier
+        MAX-ACCESS not-accessible
+        STATUS     current
+        DESCRIPTION
+             "This object uniquely identifies a conceptual row
+              in the table and reflects an assigned number used
+              to identifier a specific SIP method."
+        ::= { sipStatusCodeEntry 1 }

this new (additional) indexing object will be added to the
sipStatusCodesTable.   that able will now be indexed by
applIndex, sipStatusCodeMethod, and sipStatusCodeValue.

kevin



Bob Penfield wrote:
> The SIP MIB currently supports monitoring of response status codes, but not
> on a per method basis. I find response status code information much more
> useful when it is correlated to the method. Would it be possible to add
> this?
> 
> cheers,
> (-:bob
> 
> Robert F. Penfield
> Chief Software Architect
> Acme Packet, Inc.
> 130 New Boston Street
> Woburn, MA 01801
> bpenfield@acmepacket.com
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 



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From sip-bounces@ietf.org  Wed Oct 20 13:32:24 2004
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From: Goulet Walter-CWG009 <Walter.Goulet@motorola.com>
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Hi:

The draft 'draft-ietf-sip-connect-reuse-02' describes sharing a single SIP connection at the transport layer between two SIP endpoints. I've been trying to determine if this draft is going to be moved further along in the RFC process; currently it's in the initial ID-Exists state. I haven't been able to contact the author, hopefully he'll see this message.

Thanks,
Walter


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Use sipping@ietf.org for new developments on the application of sip
--===============1214764002==--



From sip-bounces@ietf.org  Thu Oct 21 08:28:54 2004
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Date: Wed, 20 Oct 2004 22:18:09 -0400
From: Mauricio Arango <Mauricio.Arango@Sun.COM>
Subject: Re: [Sip] verifying identity of sender
To: Dean Willis <dean.willis@softarmor.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org,
        Juha Heinanen <jh@tutpro.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>
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Folks,

I believe the approach described in draft-ietf-sip-identity-03,
based on secure exchange of authentication assurances, is the right
way to achieve end-to-end authentication among SIP entities that
don't belong to the same administrative domain.

However, I'd like to ask of why not use an already
standardized structure, specifically SAML, for the
proposed Identity header? Something similar has been
already put forward in draft-tschofenig-sip-saml, where the
'Network Asserted Identities' case in Figure 3  is
the closest to the approach in draft-ietf-sip-identity-03
(where the SAML Asserting party role is handled by
the SIP Autentication proxy).

Benefits of using SAML include:

    - Standardized and widely adopted structure for
      security (authentication, authorization, attribute) assertions.

    - Simplifies integration with other identity management systems.

    - Extensible, XML-based format - SIP-specific message information
        including headers (eg. Contact, Call-ID, CSeq)
        and the SDP body can be specified as new SAML
        attributes.

    - Transfer of attributes associated with the asserted
        identity, which enables trait-based authorization as
        described in draft-ietf-sipping-trait-authz-00.

Regards,

Mauricio Arango









Dean Willis wrote:

> Peterson, Jon wrote:
>
> >Again, Dean, my immediate concern here is whether or not this discussion
> >about a hypothetical, partially-formulated alternative is material to the
> >progression of sip-identity-03.
> >
> >
> >
> Ahah! No, this has NOTHING to do with impeding the Identity draft, as
> far as I know. But it is a suggestion of another mechanism that MIGHT be
> useful to talk about. As far as I know, it NEVER had anything to do with
> the identity draft, but was a suggestion of another technique that might
> also be useful, and deployable in the nearer term. I think we've roughly
> concluded that the specific idea suggested has little or no benefit over
> a simpler operatioal procedure, but the exercise of that analysis is
> probably something useful for the larger community.
>
> >While it is interesting to discuss alternatives that provide a weaker
> >assurance, I see no reason why the exploration of those alternatives should
> >impede the standardization of a system that seems to satisfy the broader
> >requirement.
> >
> >
>
> Me neither.
>
> >Ultimately, this thread was kicked off with the contention that acquiring
> >certificates is infeasible. Both Cullen and I provided some data about
> >certificate acquisition that seemed to argue that the problem isn't nearly
> >so bad. If you can get a publicly-verifiable certificate for under $100 a
> >year, then the expenditure is on the same order of magnitude as acquiring a
> >domain name. I haven't yet heard the parallel argument that domain names are
> >so expensive and administratively cumbersome that we should invent a new
> >alternative system for locating SIP servers, but in my opinion, this
> >contention would carry roughly the same weight as the arguments I've heard
> >against certificates so far.
> >
> >
> >
> Yes, but this thread SUGGESTED operational policies that, in the absence
> of universal certificate deployment, might cut down on sip-spam.
>
> >If there is in fact no problem with making certificates a component of the
> >identity architecture, then I'm not sure why we want to explore the
> >specification of alternatives. Alternatives are bad, when it comes to
> >security. I continue to believe that we should set a long-term goal of
> >arriving a single mandatory-to-use identity mechanism for SIP. Partial
> >solutions will not get us where we need to be.
> >
> >
> >
> Operational procedures can be put in place in the "near end", (the
> domain individual operators are operating and protecting), and
> certificates gain little utility until they happen to be widely deployed
> at the FAR end, over which said operator has little control. That being
> said, there
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


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Subject: [Sip] draft on messaging, impersonation and identity
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I've written a new draft on the design space of providing identity services
to meet the threat of impersonation in messaging systems:

http://www.ietf.org/internet-drafts/draft-peterson-message-identity-00.txt

It's a very long draft, since it tries to be a relatively complete analysis
of the problem space (though that much said, I am already aware of several
important omissions). Moreover, it tries to look at messaging at a high
level, rather than being specific to SIP. I created this draft for two
reasons, really. First, to show what the trade-offs are in the many possible
core designs of identity services that we've uncovered in our work here in
SIP. Second, I hope that it might be useful to others exploring designs for
identity mechanisms for other messaging systems.

I won't be seeking working group status for this draft here in the SIP WG,
but I thought it might be of interest. 

Jon Peterson
NeuStar, Inc.


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Date: Thu, 21 Oct 2004 13:58:20 +0300 (EEST)
Subject: Re: [Sip] verifying identity of sender
From: "Juha Heinanen" <jh@tutpro.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org,
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Dean Willis said:

> Ahah! No, this has NOTHING to do with impeding the Identity draft, as
> far as I know. But it is a suggestion of another mechanism that MIGHT be
> useful to talk about. As far as I know, it NEVER had anything to do with
> the identity draft, but was a suggestion of another technique that might
> also be useful, and deployable in the nearer term.

that is correct, but if it is so that the identity draft requires presence
of a proxy, then it needs to be fixed so that it also works in a pure
peer-to-peer environment like the mechanism that i proposed for spam
control.

-- juha


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Why do we need a Join header to be able to indicate that we want to join
an ongoing session? Wouldn't it be enough to, for the party that wants
to join, actually use the call-id for the session when sending the
join-INVITE? Suppose A and B are in a session with the dialog identified
by call-id=3D12345;tag=3DA;tag=3DB. If C now wants to join the session =
he
sends an INVITE to A or B with call-id=3D12345, from-tag=3DC, and an =
empty
to-tag. Wouldn't it make more sense that the 2 dialog-ids had some parts
in common, that is the call-id, and something that distinguished them,
that is the to and from tags? It would be much easier to find the
session related messages as they would share the same call-id.

=20

Have I missed some essential part of the Join header, or are we just
adding something that we could already express easily with what we
already have?

=20

=20

/ Christian Jansson, Hotsip


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Why do we need a Join header to be able to indicate =
that we
want to join an ongoing session? Wouldn&#8217;t it be enough to, for the =
party
that wants to join, actually use the call-id for the session when =
sending the
join-INVITE? Suppose A and B are in a session with the dialog identified =
by call-id=3D12345;tag=3DA;tag=3DB.
If C now wants to join the session he sends an INVITE to A or B with
call-id=3D12345, from-tag=3DC, and an empty to-tag. Wouldn&#8217;t it =
make more
sense that the 2 dialog-ids had some parts in common, that is the =
call-id, and
something that distinguished them, that is the to and from tags? It =
would be
much easier to find the session related messages as they would share the =
same
call-id.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Have I missed some essential part of the Join header, =
or are
we just adding something that we could already express easily with what =
we
already have?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>/ <st1:PersonName w:st=3D"on">Christian =
Jansson</st1:PersonName>,
Hotsip<o:p></o:p></span></font></p>

</div>

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mauricio Arango'" <Mauricio.Arango@Sun.COM>,
        Dean Willis
	<dean.willis@softarmor.com>
Subject: RE: [Sip] verifying identity of sender
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We're working towards a mid-term goal of unifying the existing sip-identity
work with the SIP-SAML draft. I suspect that the vehicle for this
unification while actually be the Identity-Info header rather than the
Identity header (i.e., Identity-Info could provide a means of accessing a
SAML assertion by reference).

However, I also believe it is critical that we aim towards a
universally-deployable, thinner identity assertion than SAML. SAML is heavy,
and requires a great deal of pre-association between the creator and
consumer of assertions. We need something that addresses the impersonation
problem but doesn't entail the sorts of smarts required to make SAML work.
So in any regime, I believe sip-identity will have a place; SAML would
provide supplemental information commensurate with its intrinsic
versatility.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Mauricio Arango [mailto:Mauricio.Arango@Sun.COM]
> Sent: Wednesday, October 20, 2004 10:18 PM
> To: Dean Willis
> Cc: Cullen Jennings; sip@ietf.org; Juha Heinanen; Peterson, Jon
> Subject: Re: [Sip] verifying identity of sender
> 
> 
> Folks,
> 
> I believe the approach described in draft-ietf-sip-identity-03,
> based on secure exchange of authentication assurances, is the right
> way to achieve end-to-end authentication among SIP entities that
> don't belong to the same administrative domain.
> 
> However, I'd like to ask of why not use an already
> standardized structure, specifically SAML, for the
> proposed Identity header? Something similar has been
> already put forward in draft-tschofenig-sip-saml, where the
> 'Network Asserted Identities' case in Figure 3  is
> the closest to the approach in draft-ietf-sip-identity-03
> (where the SAML Asserting party role is handled by
> the SIP Autentication proxy).
> 
> Benefits of using SAML include:
> 
>     - Standardized and widely adopted structure for
>       security (authentication, authorization, attribute) assertions.
> 
>     - Simplifies integration with other identity management systems.
> 
>     - Extensible, XML-based format - SIP-specific message information
>         including headers (eg. Contact, Call-ID, CSeq)
>         and the SDP body can be specified as new SAML
>         attributes.
> 
>     - Transfer of attributes associated with the asserted
>         identity, which enables trait-based authorization as
>         described in draft-ietf-sipping-trait-authz-00.
> 
> Regards,
> 
> Mauricio Arango
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Dean Willis wrote:
> 
> > Peterson, Jon wrote:
> >
> > >Again, Dean, my immediate concern here is whether or not 
> this discussion
> > >about a hypothetical, partially-formulated alternative is 
> material to the
> > >progression of sip-identity-03.
> > >
> > >
> > >
> > Ahah! No, this has NOTHING to do with impeding the Identity 
> draft, as
> > far as I know. But it is a suggestion of another mechanism 
> that MIGHT be
> > useful to talk about. As far as I know, it NEVER had 
> anything to do with
> > the identity draft, but was a suggestion of another 
> technique that might
> > also be useful, and deployable in the nearer term. I think 
> we've roughly
> > concluded that the specific idea suggested has little or no 
> benefit over
> > a simpler operatioal procedure, but the exercise of that analysis is
> > probably something useful for the larger community.
> >
> > >While it is interesting to discuss alternatives that 
> provide a weaker
> > >assurance, I see no reason why the exploration of those 
> alternatives should
> > >impede the standardization of a system that seems to 
> satisfy the broader
> > >requirement.
> > >
> > >
> >
> > Me neither.
> >
> > >Ultimately, this thread was kicked off with the contention 
> that acquiring
> > >certificates is infeasible. Both Cullen and I provided 
> some data about
> > >certificate acquisition that seemed to argue that the 
> problem isn't nearly
> > >so bad. If you can get a publicly-verifiable certificate 
> for under $100 a
> > >year, then the expenditure is on the same order of 
> magnitude as acquiring a
> > >domain name. I haven't yet heard the parallel argument 
> that domain names are
> > >so expensive and administratively cumbersome that we 
> should invent a new
> > >alternative system for locating SIP servers, but in my 
> opinion, this
> > >contention would carry roughly the same weight as the 
> arguments I've heard
> > >against certificates so far.
> > >
> > >
> > >
> > Yes, but this thread SUGGESTED operational policies that, 
> in the absence
> > of universal certificate deployment, might cut down on sip-spam.
> >
> > >If there is in fact no problem with making certificates a 
> component of the
> > >identity architecture, then I'm not sure why we want to explore the
> > >specification of alternatives. Alternatives are bad, when 
> it comes to
> > >security. I continue to believe that we should set a 
> long-term goal of
> > >arriving a single mandatory-to-use identity mechanism for 
> SIP. Partial
> > >solutions will not get us where we need to be.
> > >
> > >
> > >
> > Operational procedures can be put in place in the "near end", (the
> > domain individual operators are operating and protecting), and
> > certificates gain little utility until they happen to be 
> widely deployed
> > at the FAR end, over which said operator has little 
> control. That being
> > said, there
> >
> > --
> > Dean
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From sip-bounces@ietf.org  Thu Oct 21 14:00:51 2004
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Date: Thu, 21 Oct 2004 13:35:39 -0400
From: Mauricio Arango <Mauricio.Arango@Sun.COM>
To: CORVOYSIER David RD-MAPS-REN <david.corvoysier@francetelecom.com>
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Cc: "sip@ietf.org" <sip@ietf.org>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, simple@ietf.org
Subject: [Sip] Re: [Simple] Usage of Contact header in PUBLISH for
 applicationauthentication ?
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Use of TLS between the app server and the presence
server seems to assume that the PUBLISH request isn't
routed through multiple intermediate SIP proxies, which
shouldn't be a valid assumption even when both
the app server and the presence server are in the
same administrative domain.

I believe the problem of a presence server
authenticating the source of a PUBLISH request
(application server) is provided in in draft-ietf-sip-identity-03.

The other question you pose is how does the application
server can authenticate the source of presence the information
it collects. If the app server is in turn a SIMPLE presence
server, then the approach in draft-ietf-sip-identity-03
can also be applied to its presence sources. If it isn't then
a similar approach should be proposed via the use of
authentication assertions granted by an authentication provider
with which the presence sources authenticate.

Mauricio Arango


CORVOYSIER David RD-MAPS-REN wrote:

> I'm jumping into that thread because the use case described by bernhard=
 is one of those I am also struggling with (presence published on behalf =
of a presentity) ...
>
> So, from your answer I understand that TLS will authenticate the applic=
ation server towards the presence server, but does this also mean that th=
e presence server delegates user authentication to the application server=
 ?
> In that case how can the application server be sure that the user is al=
lowed to publish presence for the presentity ? I am probably missing some=
thing but it seems to me that without some kind of control a user authent=
icated by the app server could possibly publish presence for anyone ...
>
> --------------------------------
> David Corvoysier
> France Telecom R&D
> http://www.rd.francetelecom.com/
> --------------------------------
>
> > -----Message d'origine-----
> > De : simple-bounces@ietf.org [mailto:simple-bounces@ietf.org]
> > De la part de Jonathan Rosenberg
> > Envoy=E9 : jeudi 21 octobre 2004 05:24
> > =C0 : Boehmer Bernhard ICM Berlin
> > Cc : 'Aki Niemi'; SIMPLE WG
> > Objet : Re: AW: [Simple] Usage of Contact header in PUBLISH
> > for applicationa uthentication ?
> >
> > For cases where an application server is publishing, the
> > right thing is to use mutual TLS between the application
> > server and the presence server. The TLS certificates indicate
> > the identity of the element generating the publish. There is
> > no need for p-a-id here; the presence document would contain
> > the identity of the presentity. Authorization policies in the
> > presence server would typically make sure that the given
> > application server can publish state for that presentity.
> >
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple


_______________________________________________
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From sip-bounces@ietf.org  Thu Oct 21 15:40:57 2004
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From: Adam Roach <adam@nostrum.com>
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Subject: [Sip] Event Lists: Back-End Credentials
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During IESG review of draft-ietf-simple-event-list-05, the IESG raised a 
concern that section 7.1 does not specify a mandatory-to-implement 
mechanism for transfer of a secret to the RLS. This secret transfer is 
necessary under some circumstances for the purposes of authenticating on 
the users behalf for back-end subscriptions.

There is no clear-cut solution to this problem. The options that I've 
been able to think of -- and there are almost certainly more -- are as 
follows:

    * Make HTML forms mandatory

    The current example in the draft is to send the credentials in an
    HTML form over HTTP. Making this mandatory has the obvious drawback
    that it requires HTML browsers in all devices that make use of event
    lists. This is probably not an acceptable requirement.

    * Disallow back-end subscriptions altogether

    For this solution, the list server would pass back the resource list
    in notifications, along with state for any locally managed states.
    The client would be responsible for sending subscriptions for the
    state for any resources in the list that are not managed by the RLS.
    This defeats some of the purpose of event lists, since the client is
    left with maintaining several subscriptions for resources -- so it's
    really rather suboptimal.

    * Add new SIP header field (or maybe method) for credential upload

    One very simple solution would be to add a new header which contains
    a triple of [realm,userid,password]. We would specify that this
    header is disallowed except over SIPS connections. The client would
    include one or more such headers in its SUBSCRIBE request, and the
    RLS would use them to obtain information on the user's behalf.

    To make this work, we would also add a status field to the
    <instance> element that indicates something like "authorization
    failed" and the realm for which credentials are needed. The user,
    upon seeing this in the RLMI, could re-subscribe with the missing
    credentials.

    * Try to leverage ongoing work in draft-ietf-sip-identity and/or
      draft-jennings-sipping certs

    This feels like the class of problem that the ongoing identity work
    is trying to solve, but I can't figure out how to apply the existing
    drafts to it. The problem is that the RLS needs to change almost
    everything that would be protected by the identity draft.

Of these, I prefer the simplicity of adding a new SIP header-- but I'm a 
bit scared of it because the obvious, most simple solution in security 
typically ends up being the wrong solution. However, I'll point out that 
the concern that was raised by the IESG was *not* that we are passing 
the user's secret to a third party; it was the fact that we're not 
defining *how* to do so.

Unless I'm missing something, the RLS simply needs to be trusted to act 
on the user's behalf to make more-or-less arbitrary SUBSCRIBE requests, 
since the user doesn't know the contents of the list before subscribing. 
There might be some really clever way to do this without passing the 
user's secret over wholesale, but I don't see it.

Thoughts? Ideas? Brilliant solutions?

/a

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Thu Oct 21 20:41:36 2004
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] Event Lists: Back-End Credentials
Date: Thu, 21 Oct 2004 16:00:55 -0400
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Sigh. Delegation is hard. I frankly don't like any of these solutions 
because they allow full user impersonation by the RLS, which is a lot 
more trust that I think ought to be given it.

If we want to really tackle delegation, I'd suggest that we extend SIP 
generally to allow SIP-specific delegation capability. One way to do 
this is for the user to construct a certificate for the RLS giving it 
specific rights (e.g. Subscribe on my behalf), and then signing it 
based on a credential that would be accepted by the ultimate target of 
the request. SAML might be our friend here.

It might be worth going back and seeing what applies to REFER as well 
while we're at it since REFER has some pretty brittle security 
properties due to our needing to to get done quickly.

Dave.

On Oct 21, 2004, at 3:24 PM, Adam Roach wrote:

> During IESG review of draft-ietf-simple-event-list-05, the IESG raised 
> a concern that section 7.1 does not specify a mandatory-to-implement 
> mechanism for transfer of a secret to the RLS. This secret transfer is 
> necessary under some circumstances for the purposes of authenticating 
> on the users behalf for back-end subscriptions.
>
> There is no clear-cut solution to this problem. The options that I've 
> been able to think of -- and there are almost certainly more -- are as 
> follows:
>
>    * Make HTML forms mandatory
>
>    The current example in the draft is to send the credentials in an
>    HTML form over HTTP. Making this mandatory has the obvious drawback
>    that it requires HTML browsers in all devices that make use of event
>    lists. This is probably not an acceptable requirement.
>
>    * Disallow back-end subscriptions altogether
>
>    For this solution, the list server would pass back the resource list
>    in notifications, along with state for any locally managed states.
>    The client would be responsible for sending subscriptions for the
>    state for any resources in the list that are not managed by the RLS.
>    This defeats some of the purpose of event lists, since the client is
>    left with maintaining several subscriptions for resources -- so it's
>    really rather suboptimal.
>
>    * Add new SIP header field (or maybe method) for credential upload
>
>    One very simple solution would be to add a new header which contains
>    a triple of [realm,userid,password]. We would specify that this
>    header is disallowed except over SIPS connections. The client would
>    include one or more such headers in its SUBSCRIBE request, and the
>    RLS would use them to obtain information on the user's behalf.
>
>    To make this work, we would also add a status field to the
>    <instance> element that indicates something like "authorization
>    failed" and the realm for which credentials are needed. The user,
>    upon seeing this in the RLMI, could re-subscribe with the missing
>    credentials.
>
>    * Try to leverage ongoing work in draft-ietf-sip-identity and/or
>      draft-jennings-sipping certs
>
>    This feels like the class of problem that the ongoing identity work
>    is trying to solve, but I can't figure out how to apply the existing
>    drafts to it. The problem is that the RLS needs to change almost
>    everything that would be protected by the identity draft.
>
> Of these, I prefer the simplicity of adding a new SIP header-- but I'm 
> a bit scared of it because the obvious, most simple solution in 
> security typically ends up being the wrong solution. However, I'll 
> point out that the concern that was raised by the IESG was *not* that 
> we are passing the user's secret to a third party; it was the fact 
> that we're not defining *how* to do so.
>
> Unless I'm missing something, the RLS simply needs to be trusted to 
> act on the user's behalf to make more-or-less arbitrary SUBSCRIBE 
> requests, since the user doesn't know the contents of the list before 
> subscribing. There might be some really clever way to do this without 
> passing the user's secret over wholesale, but I don't see it.
>
> Thoughts? Ideas? Brilliant solutions?
>
> /a
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


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Date: Thu, 21 Oct 2004 19:52:32 -0400
From: Mauricio Arango <Mauricio.Arango@Sun.COM>
Subject: Re: [Sip] verifying identity of sender
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Thanks for the update on unifications of the sip-identity
and sip-saml drafts. Additional commments inline:

"Peterson, Jon" wrote:

> We're working towards a mid-term goal of unifying the existing sip-identity
> work with the SIP-SAML draft. I suspect that the vehicle for this
> unification while actually be the Identity-Info header rather than the
> Identity header (i.e., Identity-Info could provide a means of accessing a
> SAML assertion by reference).

This scheme would work, it's actually the SAML IdP (Identity Provider)
artifact binding approach.

>
>
> However, I also believe it is critical that we aim towards a
> universally-deployable, thinner identity assertion than SAML. SAML is heavy,
> and requires a great deal of pre-association between the creator and

For clarification, is the comment of "SAML heavy" because of the
pre-associations only?

>
> consumer of assertions. We need something that addresses the impersonation
> problem but doesn't entail the sorts of smarts required to make SAML work.
> So in any regime, I believe sip-identity will have a place; SAML would
> provide supplemental information commensurate with its intrinsic
> versatility.

Sounds like a good initial approach.

Mauricio Arango



>
>
> Jon Peterson
> NeuStar, Inc.
>
> > -----Original Message-----
> > From: Mauricio Arango [mailto:Mauricio.Arango@Sun.COM]
> > Sent: Wednesday, October 20, 2004 10:18 PM
> > To: Dean Willis
> > Cc: Cullen Jennings; sip@ietf.org; Juha Heinanen; Peterson, Jon
> > Subject: Re: [Sip] verifying identity of sender
> >
> >
> > Folks,
> >
> > I believe the approach described in draft-ietf-sip-identity-03,
> > based on secure exchange of authentication assurances, is the right
> > way to achieve end-to-end authentication among SIP entities that
> > don't belong to the same administrative domain.
> >
> > However, I'd like to ask of why not use an already
> > standardized structure, specifically SAML, for the
> > proposed Identity header? Something similar has been
> > already put forward in draft-tschofenig-sip-saml, where the
> > 'Network Asserted Identities' case in Figure 3  is
> > the closest to the approach in draft-ietf-sip-identity-03
> > (where the SAML Asserting party role is handled by
> > the SIP Autentication proxy).
> >
> > Benefits of using SAML include:
> >
> >     - Standardized and widely adopted structure for
> >       security (authentication, authorization, attribute) assertions.
> >
> >     - Simplifies integration with other identity management systems.
> >
> >     - Extensible, XML-based format - SIP-specific message information
> >         including headers (eg. Contact, Call-ID, CSeq)
> >         and the SDP body can be specified as new SAML
> >         attributes.
> >
> >     - Transfer of attributes associated with the asserted
> >         identity, which enables trait-based authorization as
> >         described in draft-ietf-sipping-trait-authz-00.
> >
> > Regards,
> >
> > Mauricio Arango
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > Dean Willis wrote:
> >
> > > Peterson, Jon wrote:
> > >
> > > >Again, Dean, my immediate concern here is whether or not
> > this discussion
> > > >about a hypothetical, partially-formulated alternative is
> > material to the
> > > >progression of sip-identity-03.
> > > >
> > > >
> > > >
> > > Ahah! No, this has NOTHING to do with impeding the Identity
> > draft, as
> > > far as I know. But it is a suggestion of another mechanism
> > that MIGHT be
> > > useful to talk about. As far as I know, it NEVER had
> > anything to do with
> > > the identity draft, but was a suggestion of another
> > technique that might
> > > also be useful, and deployable in the nearer term. I think
> > we've roughly
> > > concluded that the specific idea suggested has little or no
> > benefit over
> > > a simpler operatioal procedure, but the exercise of that analysis is
> > > probably something useful for the larger community.
> > >
> > > >While it is interesting to discuss alternatives that
> > provide a weaker
> > > >assurance, I see no reason why the exploration of those
> > alternatives should
> > > >impede the standardization of a system that seems to
> > satisfy the broader
> > > >requirement.
> > > >
> > > >
> > >
> > > Me neither.
> > >
> > > >Ultimately, this thread was kicked off with the contention
> > that acquiring
> > > >certificates is infeasible. Both Cullen and I provided
> > some data about
> > > >certificate acquisition that seemed to argue that the
> > problem isn't nearly
> > > >so bad. If you can get a publicly-verifiable certificate
> > for under $100 a
> > > >year, then the expenditure is on the same order of
> > magnitude as acquiring a
> > > >domain name. I haven't yet heard the parallel argument
> > that domain names are
> > > >so expensive and administratively cumbersome that we
> > should invent a new
> > > >alternative system for locating SIP servers, but in my
> > opinion, this
> > > >contention would carry roughly the same weight as the
> > arguments I've heard
> > > >against certificates so far.
> > > >
> > > >
> > > >
> > > Yes, but this thread SUGGESTED operational policies that,
> > in the absence
> > > of universal certificate deployment, might cut down on sip-spam.
> > >
> > > >If there is in fact no problem with making certificates a
> > component of the
> > > >identity architecture, then I'm not sure why we want to explore the
> > > >specification of alternatives. Alternatives are bad, when
> > it comes to
> > > >security. I continue to believe that we should set a
> > long-term goal of
> > > >arriving a single mandatory-to-use identity mechanism for
> > SIP. Partial
> > > >solutions will not get us where we need to be.
> > > >
> > > >
> > > >
> > > Operational procedures can be put in place in the "near end", (the
> > > domain individual operators are operating and protecting), and
> > > certificates gain little utility until they happen to be
> > widely deployed
> > > at the FAR end, over which said operator has little
> > control. That being
> > > said, there
> > >
> > > --
> > > Dean
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Dean Willis writes:
 > 
 > Wearing my chair hat and thinking about draft-ietf-sip-identity-03, 
 > which IS an approved working-group effort that WAS scheduled to be 
 > finished last April . . .
 > 
 >  From the minutes at IETF 60: "It was observed that few participants in 
 > the room displayed confidence in their understanding of the problem or 
 > the proposal, and that further study would be a Good Thing."
 > 
 > Since then, we have talked a lot about identity-related issues on the list.
 > 
 > We've hashed through some of the less-understood issues, including the 
 > tremendous similarity between the SIP spam problem and the SMTP spam 
 > problem, and Michael Thomas pointed us at the MASS work, which is 
 > astoundingly similar to sip-identity.

>From my poor understanding, the principle
difference is what the trust anchor is. That has a
tendency to either make or break these things,
especially in the large... which is what led us to
our use of DNS as a somewhat more "conservative"
compromise of deployability over security.

But! The question that I really have is whether
anybody is implementing and especially deploying
this, and if so, obviously how it's going.  We're
getting a lot of interest on the MASS front due to
the dire situation with email which may in fact
move some of these immovable objects, so unless
people are actually making use of this, waiting to
see what happens may not be a terrible option.

 > Jon and Cullen proved that real CS-signed certificates are readily 
 > available at a cost not significantly different than that of a domain 
 > registration, or slightly less than the price of a decent bottle of 
 > Scotch, for some definition of the word "decent".

There's more to it than money, of course; there's
institutional, um, considerations too. And PKI's
don't have a very encouraging track record beyond
web stuff; whether it's reasonable being a
different question than whether it is.

		   Mike


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David R Oran wrote:

> Sigh. Delegation is hard. I frankly don't like any of these solutions 
> because they allow full user impersonation by the RLS, which is a lot 
> more trust that I think ought to be given it.


I'm not too crazy about them either (I'll note that suggestion #2 
doesn't have that property, but at the cost of making the protocol far 
less useful).

>
> If we want to really tackle delegation, I'd suggest that we extend SIP 
> generally to allow SIP-specific delegation capability. One way to do 
> this is for the user to construct a certificate for the RLS giving it 
> specific rights (e.g. Subscribe on my behalf), and then signing it 
> based on a credential that would be accepted by the ultimate target of 
> the request. SAML might be our friend here.


That feels a lot like what I was trying to get at with my suggestion #4, 
albeit with a slightly different mechanism. The problem is that pursuing 
this kind of work will be very time consuming... and the event list 
document has been in limbo for over a year already. I'm not necessarily 
advocating expediency over security -- it's been so long since I started 
this work that I'm half inclined to cripple it (by removing back-end 
subscriptions) rather than wait several years for a general purpose 
solution to the delegation problem. Not that I *want* to do this -- it 
would remove most of the utility of the extension for the 3G guys -- but 
I do want to put this work to bed.

I'm not sure if it represents a reasonable compromise, but I'm not 
averse to putting in the "use this header to transfer your secret" with 
a strongly worded note indicating the implications if you do, and an 
indication that future mechanisms may define more constrained ways to 
delegate limited subsets of authority to another node (possibly for a 
constrained time period).

/a

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_______________________________________________
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From: Boehmer Bernhard ICM Berlin <bernhard.boehmer@siemens.com>
To: "'David R Oran'" <oran@cisco.com>, Adam Roach <adam@nostrum.com>
Subject: AW: [Sip] Event Lists: Back-End Credentials
Date: Fri, 22 Oct 2004 14:35:17 +0200
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Cc: sip@ietf.org, "'Mauricio.Arango@Sun.COM'" <Mauricio.Arango@Sun.COM>,
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Hi,
I think controlling whether an RLS (or in a general any service)=20
can perform certain subscriptions or publications (in case of=20
presence) should be controlled by the notifier/PUBLISH-recipient.
Thus, additionally to the control whether a certain *user* is allowed =
to
subscribe/publish the notifier/PUBLISH recipient should be also able
to control whether an *application* is allowed to act on behalf of the
user. Currently, the needed application authentication is under =
discussion
(see the SIMPLE thread=20
"[Simple] Usage of Contact header in PUBLISH for application =
authentication?").
Mauricio also mentioned the identity draft in this thread but I haven't =
yet=20
discovered how this drafts solves this issue.
		Bernhard



> -----Urspr=FCngliche Nachricht-----
> Von: David R Oran [mailto:oran@cisco.com]=20
> Gesendet: Donnerstag, 21. Oktober 2004 22:01
> An: Adam Roach
> Cc: sip@ietf.org
> Betreff: Re: [Sip] Event Lists: Back-End Credentials
>=20
>=20
> Sigh. Delegation is hard. I frankly don't like any of these solutions =

> because they allow full user impersonation by the RLS, which is a lot =

> more trust that I think ought to be given it.
>=20
> If we want to really tackle delegation, I'd suggest that we=20
> extend SIP=20
> generally to allow SIP-specific delegation capability. One way to do=20
> this is for the user to construct a certificate for the RLS giving it =

> specific rights (e.g. Subscribe on my behalf), and then signing it=20
> based on a credential that would be accepted by the ultimate=20
> target of=20
> the request. SAML might be our friend here.
>=20
> It might be worth going back and seeing what applies to REFER as well =

> while we're at it since REFER has some pretty brittle security=20
> properties due to our needing to to get done quickly.
>=20
> Dave.
>=20
> On Oct 21, 2004, at 3:24 PM, Adam Roach wrote:
>=20
> > During IESG review of draft-ietf-simple-event-list-05, the=20
> IESG raised=20
> > a concern that section 7.1 does not specify a=20
> mandatory-to-implement=20
> > mechanism for transfer of a secret to the RLS. This secret=20
> transfer is=20
> > necessary under some circumstances for the purposes of=20
> authenticating=20
> > on the users behalf for back-end subscriptions.
> >
> > There is no clear-cut solution to this problem. The options=20
> that I've=20
> > been able to think of -- and there are almost certainly=20
> more -- are as=20
> > follows:
> >
> >    * Make HTML forms mandatory
> >
> >    The current example in the draft is to send the credentials in =
an
> >    HTML form over HTTP. Making this mandatory has the=20
> obvious drawback
> >    that it requires HTML browsers in all devices that make=20
> use of event
> >    lists. This is probably not an acceptable requirement.
> >
> >    * Disallow back-end subscriptions altogether
> >
> >    For this solution, the list server would pass back the=20
> resource list
> >    in notifications, along with state for any locally=20
> managed states.
> >    The client would be responsible for sending subscriptions for =
the
> >    state for any resources in the list that are not managed=20
> by the RLS.
> >    This defeats some of the purpose of event lists, since=20
> the client is
> >    left with maintaining several subscriptions for=20
> resources -- so it's
> >    really rather suboptimal.
> >
> >    * Add new SIP header field (or maybe method) for=20
> credential upload
> >
> >    One very simple solution would be to add a new header=20
> which contains
> >    a triple of [realm,userid,password]. We would specify that this
> >    header is disallowed except over SIPS connections. The=20
> client would
> >    include one or more such headers in its SUBSCRIBE=20
> request, and the
> >    RLS would use them to obtain information on the user's behalf.
> >
> >    To make this work, we would also add a status field to the
> >    <instance> element that indicates something like "authorization
> >    failed" and the realm for which credentials are needed. The =
user,
> >    upon seeing this in the RLMI, could re-subscribe with the =
missing
> >    credentials.
> >
> >    * Try to leverage ongoing work in draft-ietf-sip-identity and/or
> >      draft-jennings-sipping certs
> >
> >    This feels like the class of problem that the ongoing=20
> identity work
> >    is trying to solve, but I can't figure out how to apply=20
> the existing
> >    drafts to it. The problem is that the RLS needs to change almost
> >    everything that would be protected by the identity draft.
> >
> > Of these, I prefer the simplicity of adding a new SIP=20
> header-- but I'm=20
> > a bit scared of it because the obvious, most simple solution in=20
> > security typically ends up being the wrong solution. However, I'll=20
> > point out that the concern that was raised by the IESG was=20
> *not* that=20
> > we are passing the user's secret to a third party; it was the fact=20
> > that we're not defining *how* to do so.
> >
> > Unless I'm missing something, the RLS simply needs to be trusted to =

> > act on the user's behalf to make more-or-less arbitrary SUBSCRIBE=20
> > requests, since the user doesn't know the contents of the=20
> list before=20
> > subscribing. There might be some really clever way to do=20
> this without=20
> > passing the user's secret over wholesale, but I don't see it.
> >
> > Thoughts? Ideas? Brilliant solutions?
> >
> > /a
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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To: hongchuan.yuan@263.net, sip@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] Location Header
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At 02:56 PM 10/22/2004 +0800, hongchuan.yuan@263.net wrote:

>hello,everyone
>can you give some information about the current status of "location"=20
>header in SIP.

There is no location "header" in SIP, but a location MIME body part effort=
=20
in the SIPPING WG - which is based on the PIDF-LO that was defined in the=20
Geopriv WG. The SIPPING effort is being defined  in the ID:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-location-requirements=
-01.txt

The above effort will be updated in the next few days to version -02 and=20
announced on the SIPPING list.

BTW - the PIDF-LO ID from Geopriv is at:

http://www.ietf.org/internet-drafts/draft-ietf-geopriv-pidf-lo-03.txt

and is in the RFC-Editor now.


>thans.
>
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D 263=B5=E7=D7=D3=D3=CA=BC=FE=A3=AD=D0=C5=C0=B5=D3=CA=D7=D4=D7=A8=D2=B5
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: [Sip] RFC 3911 on The Session Initiation Protocol (SIP) "Join"
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--NextPart


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


        RFC 3911

        Title:      The Session Initiation Protocol (SIP)
                    "Join" Header
        Author(s):  R. Mahy, D. Petrie
        Status:     Standards Track
        Date:       October 2004
        Mailbox:    rohan@airespace.com, dpetrie@pingtel.com
        Pages:      17
        Characters: 35373
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-join-03.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3911.txt


This document defines a new header for use with SIP multi-party
applications and call control.  The Join header is used to logically
join an existing SIP dialog with a new SIP dialog.  This primitive
can be used to enable a variety of features, for example: "Barge-In",
answering-machine-style "Message Screening" and "Call Center
Monitoring".  Note that definition of these example features is
non-normative.

This document is a product of the Session Initiation Protocol 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
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USC/Information Sciences Institute

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_______________________________________________
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From sip-bounces@ietf.org  Fri Oct 22 18:39:17 2004
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Date: Fri, 22 Oct 2004 16:34:31 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
To: sip@ietf.org
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Organization: MCI
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Cc: "'Rohan Mahy'" <rohan@rohan.com>,
        "'Dean Willis'" <dwillis@dynamicsoft.com>,
        "'Peterson, Jon'" <jon.peterson@neustar.biz>, mankin@psg.com
Subject: [Sip] draft-ietf-sip-identity-03.txt
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The I-D on "Certificate Management Service for SIP" (
http://www.ietf.org/internet-drafts/draft-ietf-sipping-certs-00.txt )
addresses a problem we have been struggling with for some time. It is the
very approach that makes sense IMHO. 

I propose we put it on the agenda to discuss and make it a WG item.

Thanks, Henry

Henry Sinnreich
MCI
1201 E. Arapaho Rd./Office 1020
Richardson, Texas 75081
USA




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From sip-bounces@ietf.org  Fri Oct 22 18:54:07 2004
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I have just submitted a new version of the event list draft for 
publication. Until it appears in the archives, you can retrieve a copy from:

http://www.nostrum.com/~adam/ids/draft-ietf-simple-event-list-06.txt

/a

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From sip-bounces@ietf.org  Fri Oct 22 19:00:27 2004
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> -----Original Message-----
> From: Mauricio Arango [mailto:Mauricio.Arango@Sun.COM]
> 
> For clarification, is the comment of "SAML heavy" because of the
> pre-associations only?

Pre-association and federation, agreement on various SAML attribute and so
on is one aspect of it. I also think the support for XML, web services, and
so on cannot be universally assumed in a SIP environment - a widespread
mandate for SAML support would entail a lot of additional development,
beyond SIP's current implementation challenge.

Our emphasis in identity design has been on providing a very light solution,
something that doesn't require us to diverge significantly from the
operational models and implementation already described in the core
standard. We also want to provide something for identity that is universal,
and useful in essentially all contexts that SIP is used. I don't think we
can position SAML to play that part in SIP's architecture, though it can
certainly provide valuable supplemental security assertions for some
deployments.

Jon Peterson
NeuStar, Inc.

> 
> Sounds like a good initial approach.
> 
> Mauricio Arango
> 
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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Cc: sip@ietf.org, "'Rohan Mahy'" <rohan@rohan.com>,
        "'Peterson,
	Jon'" <jon.peterson@neustar.biz>, mankin@psg.com
Subject: [Sip] Re: draft-ietf-sip-identity-03.txt
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Henry Sinnreich wrote:

>The I-D on "Certificate Management Service for SIP" (
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-certs-00.txt )
>addresses a problem we have been struggling with for some time. It is the
>very approach that makes sense IMHO. 
>
>I propose we put it on the agenda to discuss and make it a WG item.
>
>  
>

It is now a working group item, and I expect we'll be talking about it.

--
Dean



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Date: Mon, 25 Oct 2004 16:38:29 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: FW: [Sip] Proposed bug fix to RFC3261 arising from draft-elwell-s
	ip-state-u pdate-02
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I received no responses to the following mail, asking for opinions on which
direction to take. The default is, of course, option 1 (do nothing - no
clarifications to state update will be made). Therefore in the absence of
any proposals to the contrary by the Washington meeting, I will drop this
work.

John Elwell (john.elwell@siemens.com)

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Elwell, John
> Sent: 01 October 2004 09:10
> To: 'Paul Kyzivat'
> Cc: 'sip@ietf.org'
> Subject: RE: [Sip] Proposed bug fix to RFC3261 arising from 
> draft-elwell-s ip-state-u pdate-02
> 
> Paul,
> 
> Thanks for your comment. You have made it clear before that there is a
> distinction between call state and dialog state, and that there is not
> always a 1:1 relationship between them.
> 
> However, there are moves to deprecate dialog re-use. So 
> although there are
> examples of dialog re-use today (the REFER case being the 
> prime example), I
> don't anticipate new applications of this technique to be 
> endorsed and there
> is likely to be a move to phase out existing forms of re-use. 
> 
> Moreover there cannot be more than one call per dialog. Therefore if a
> dialog supports a call, there is not a lot of difference in principle
> between dialog state and call state, although if the dialog 
> also supports a
> subscription, call state information will not survive after a 
> BYE, even
> though dialog state will survive if the subscription survives.
> 
> With these considerations in mind, perhaps we don't need to 
> put an excessive
> amount of effort into distinguishing between call state and 
> dialog state.
> Call state just becomes a part of dialog state, existing on 
> dialogs that
> support calls but not on dialogs that support subscriptions.
> 
> I eliminated some of the elements such as Call-Info, the 
> updating of which
> did not appear to be of great interest to anyone and which, 
> as you say, are
> call state. Yes, session information I guess is also call state.
> 
> I forgot to add Required/Supported/Allowed headers, which I 
> guess are in
> fact part of the dialog state. From other specifications, PAI 
> is an example
> of data that is part of dialog state.
> 
> I guess the main question we need to answer is whether this proposed
> clarification to RFC3261 (and eventually RFC3311) should 
> embrace dialog
> state change, call state change or both. One difficulty is 
> that the term
> "call" is not well-defined in RFC3261 (the Call-Id header 
> unfortunately is
> dialog state, not call state). Then the question arises 
> whether we should
> embark on trying to define the term call and hence call state.
> 
> So I see the following options, and I would like some 
> feedback from the
> list:
> 
> 1. Do nothing.
> 
> 2. Do not attempt to distinguish between call state and 
> dialog state, on the
> basis that the desire is to phase out dialog re-use (some 
> dialogs will of
> course also be calls, but a dialog should not serve two purposes:
> subscription and call). In other words, continue as proposed 
> in my previous
> email (possibly restoring things like Call-Info if there is interest).
> 
> 3. Stick rigidly to dialog state (so session information 
> would be removed
> from my proposal).
> 
> 4. Define "call" and "call state" and distinguish them from 
> "dialog" and
> "dialog state". Handle both call state update and dialog state update.
> 
> Opinions please.
> 
> John Elwell (john.elwell@siemens.com)
> 
> Paul Kyzivat wrote:
> I agree with what you are trying to do. But one rather significant 
> detail with what you propose is that I don't believe the session 
> description (SDP) is dialog state. It is call state. (A dialog that 
> exists for the purpose of a SUBSCRIBE would not have this.)
> 
> That in some sense guts this proposal, since now there are no 
> examples 
> in this document of additional dialog state. However I still 
> think the 
> clarification is needed.
> 
> In addition, 3261 sweeps the distinction between dialog state 
> and call 
> state under the rug. Once alternative uses of dialogs are in 
> play it is 
> important to make this distinction clear, and to identify those items 
> that are part of call state. This includes the session 
> description info 
> (the negotiated offer and answer as well as potentially 
> another proposed 
> change to offer and answer). It also includes the data 
> associated with 
> session timer, and possibly Call-Info, Alert-Info, Subject. (Which if 
> any of these is truly call state remains to be discussed.)
> 
> 	Paul
> 
> Elwell, John wrote:
> > At the SIP meeting in San Diego, I was asked to raise on 
> the mailing list
> > the individual bug fixes proposed in the state-update 
> draft, so that each
> > can be given due scrutiny. I decided to start with RFC3261. 
> Here only a
> few
> > changes are proposed and it does not seem sensible to consider these
> > separately, so I propose them here as a single bug fix. If 
> we can gain
> > consensus on those, I will follow it up with messages 
> proposing bug fixes
> to
> > the other documents (mainly RFC3311).
> > 
> > As a result of comments by Paul Kyzivat raised on the list 
> recently, I
> have
> > modified the proposed changes slightly. In particular, I 
> have dropped any
> > headers that are not obviously dialog state changes, 
> including those that
> > might not be considered state at all (e.g., Alert-Info) and 
> those that
> might
> > be considered call state rather than dialog state (e.g., Call-Info,
> > Subject). If anyone wants these re-instating, please say so.
> > 
> > Proposed bug fix to RFC3261:
> > 
> > Add the following paragraph to the end of 12 (prior to 12.1):
> > 
> > "A dialog can also have certain other state information 
> that can change
> > during the lifetime of the dialog. This includes the 
> session description
> (as
> > negotiated by SDP offer/answer), information from certain 
> other headers
> and
> > information from certain other bodies. There are no 
> examples from this
> RFC.
> > However, extensions may define other headers or bodies containing
> > information that contributes to the state of a dialog and can change
> during
> > the lifetime of the dialog."
> > 
> > Add two new paragraphs to the end of 12.2 (prior to 12.2.1):
> > 
> > "A request, whether or not it is a target refresh request, 
> MAY update
> > certain other information transmitted at the time of dialog 
> establishment.
> > Although there are no examples in this RFC, other 
> extensions may define
> > headers or bodies that are appropriate for updating during a dialog.
> > 
> > "In this RFC, the only request defined that is suitable for updating
> > information during a dialog is re-INVITE (see section 14). Other
> extensions
> > may define different requests suitable for updating 
> information during a
> > dialog."
> > 
> > Add to the end of the first paragraph of 14.1:
> > 
> > "A UAC MAY send a session description that is unchanged, if 
> the purpose of
> > the re-INVITE is solely for updating dialog-related information."
> > 
> > John Elwell (john.elwell@siemens.com)
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Michael Thomas'" <mat@cisco.com>,
        Dean Willis <dean.willis@softarmor.com>
Subject: RE: [Sip] Discussion on  draft-ietf-sip-identity-03.txt
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Michael,

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
<snip>
>  > problem, and Michael Thomas pointed us at the MASS work, which is 
>  > astoundingly similar to sip-identity.
> 
> >From my poor understanding, the principle
> difference is what the trust anchor is. That has a
> tendency to either make or break these things,
> especially in the large... which is what led us to
> our use of DNS as a somewhat more "conservative"
> compromise of deployability over security.
> 

Right. Just to articulate to the group what that distinction is, IIM
(Michael's proposal in MASS) uses the records in the DNS to point to a key
store where you can find (uncertified) public keys for a particular user.

Certainly, I personally wouldn't recommend that you use CA-issued
certificates for IIM, precisely because your mechanism manages keys on a
per-user basis. Getting PKI to end users (in terms of enrollment and
credential distribution) is very tricky. Hence, using uncertified or
self-signed keys for your architecture makes more sense. We do that for
end-users in SIP, too (in the jennings-certs draft).

That much said, it is much harder to prove that you are a user than to prove
that you are a domain. In your architecture, the DNS is used to bind
credentials to the issuing domain rather than a certificate. If, say, you
required people to access your key store via HTTPS, then there would be
little difference between the trust anchors of our approach. I'm surprised
that this doesn't even merit a MAY in your proposal; while one could argue
that the DNS is sufficiently secure, it is certainly not unrealistic or
operationally challenging to use HTTPS.

> But! The question that I really have is whether
> anybody is implementing and especially deploying
> this, and if so, obviously how it's going.  We're
> getting a lot of interest on the MASS front due to
> the dire situation with email which may in fact
> move some of these immovable objects, so unless
> people are actually making use of this, waiting to
> see what happens may not be a terrible option.
> 

I understand that there are some implementations underway, but I don't know
about deployments.

While my high-level message-identity draft shows the higher layer
commonality between the identity needs of SIP and email, it also throws a
lot of the differences into relief. After going through that exercise, I'm
not sure I would make the same recommendations for SIP and email identity.
But of course, it wasn't the point of the draft to show that MASS should use
sip-identity or that SIP should use DomainKeys or IIM or what have you - it
was to show a common design space in which the trade-offs are clear.

>  > Jon and Cullen proved that real CS-signed certificates are readily 
>  > available at a cost not significantly different than that of a domain 
>  > registration, or slightly less than the price of a decent bottle of 
>  > Scotch, for some definition of the word "decent".
> 
> There's more to it than money, of course; there's
> institutional, um, considerations too. And PKI's
> don't have a very encouraging track record beyond
> web stuff; whether it's reasonable being a
> different question than whether it is.
> 

You and I are in perfect agreement that the traditional manner of leveraging
PKI to support messaging has proved inadequate. Of course, I take exception
to the idea that sip-identity relies on certificates in that manner - it is
not predicated on enrolling individual users of the system and distributing
certified credentials to these users. In fact, it uses certificates the same
way that "web stuff" does - servers have 'em, users and other servers
validate 'em. This is a tried and tested approach to enrollment into PKI,
and tons of commercial services make these certificates available. I'm not
sure what limiting institutional considerations you have mind.

Jon Peterson
NeuStar, Inc.

> 		   Mike
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: An Extension to the Session Initiation Protocol for Request History Information
	Author(s)	: M. Barnes
	Filename	: draft-ietf-sip-history-info-04.txt
	Pages		: 47
	Date		: 2004-10-25
	
This draft defines a standard mechanism for capturing the history 
   information associated with a SIP request.  This capability enables 
   many enhanced services by providing the information as to how and why 
   a call arrives at a specific application or user.  This draft defines 
   a new optional SIP header, History-Info, for capturing the history 
   information in requests. A new option tag, Histinfo, to be included 
   in the Supported header, is defined to allow UAs to indicate whether 
   the History-Info should be returned in responses to a request which 
   has captured the history information. A new priv-value, history, is 
   added to the Privacy header to allow for privacy handling of the 
   History-Info header.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.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 Session Initiation Protocol Working Group of the IETF.

	Title		: Connection Reuse in the Session Initiation Protocol (SIP)
	Author(s)	: R. Mahy
	Filename	: draft-ietf-sip-connect-reuse-03.txt
	Pages		: 14
	Date		: 2004-10-25
	
When SIP entities use a connection oriented protocol to send a
request, they typically originate their connections from an ephemeral
port. The SIP protocol includes mechanisms which insure that
responses to a request, and new requests sent in the original
direction reuse an existing connection.  However, new requests sent
in the opposite direction are unlikely to reuse the existing
connection.  This frequently causes a pair of SIP entities to use one
connection for requests sent in each direction, and can result in
potential scaling and performance problems.  This document proposes
requirements and a mechanism which address this deficiency.

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Piers O'Hanlon wrote:
> Hi,
> 
> I've taken a look at the draft and have the following comments:

Thanks for your comments.

> 
> General comments: - The fact that it isn't mandatory to mark all
> responses with RP headers means that one should at least make it a
> SHOULD(MUST?) to provide "resource prioritization" based on a dialog
> or session initiated with the RP header (possibly requiring Stateful
> proxy?). Otherwise it seems that being able to know if a node
> "supports Resource priority" (using OPTIONS or a Require hdr) may not
> be of use if you can't be assured the whole dialog/session will be
> assured of the "resource prioritization"? If it's not specified in
> this document then I guess it will need to be specified in some
> external policy document...(out of scope)

There are two types of resource prioritization, as alluded to in the 
document (Introduction), namely to prioritize the signaling handling 
itself and, more importantly, the resources such signaling controls, 
e.g., trunk circuits.

Resources such as trunk circuits are inherently stateful, as they are at 
UAs, and thus don't require the response to carry the R-P indication.

The only time the response would need to carry the R-P header is for 
prioritizing the handling of the SIP request itself. Since a response is 
part of a transaction, transaction-stateful proxies (i.e., most of them) 
would already know that the transaction has high priority and can handle 
this without additional indication. Thus, only the rare case of a 
stateless proxy is relevant. However, since that proxy can't readily 
ascertain that the response is authorized (ignoring S/MIME for now), 
this isn't likely to be useful.

I'll add a sentence that summarizes this.


> 
> 
> Section 3.3 - minor comment: It might be useful to label table
> something like 'Table 1'. 

Done.

- In the second the paragraph the 420
> response is incorrectly defined as "(Not Supported)" - when it should
> be "(Bad Extension)". 

Fixed.

- In the same paragraph the explanation
> following the handling of the 420 response seems at odds with the
> description in section 4.2.2 and that of RFC3261. Firstly this is
> because it indicates that a 420 is generated when it encounters a
> unknown RP value (on an RP capable node) -  shouldn't that be covered
> in section 4.2.2 

That part of the paragraph below the table is an orphaned remnant of 
some earlier behavioral description. The error behavior is now described 
in Section 4.2. I ditched the confusing set of sentences.

- Shouldn't a 420 be generated only by non RP
> capable node in response to Require? Secondly it seems to reccomend
> adding an 'Accept-Resource-Priority' header when RFC3261 specifies
> adding an "Unsupported" header for a 420 response? Or maybe it was
> intended to talk about the 417 handling here?

See above.


> 
> Example 7.2 - In this example - according to section 4.2.2 a 417
> response should only generated when in strict mode, in which case the
> shouldn't the INVITE contain the Require header?

Indeed.

> 
> 
> Piers O'Hanlon --------------_____________ Computer Science
> Department University College London

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From sip-bounces@ietf.org  Tue Oct 26 02:08:57 2004
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After a discussion with Dean about some of the issues surrounding the 
back-end credential issue that I raised earlier on the list, I struck 
upon a solution that might get us past this logjam.

The lynch pin to this solution is that the RLS does not ever pretend to 
be the user. Instead, it always subscribes as its own identity. It can 
present credentials as necessary to authenticate itself, but only as 
itself, not as its subscribers.

A consequence of this is that, lacking any other mechanism, the 
resources in a resource list for which the RLS is not authoritative will 
contain only the information that their notifier is willing to reveal to 
the RLS -- which is generally the same information they would render to 
unauthenticated users. In some cases, this might result in no useful 
information being transferred.

Of course, the resource would still be present in the RLMI document; 
this gives the subscriber the *option* to directly subscribe to any 
resources for which the RLS is not authoritative. To facilitate the 
client's ability to perform such subscriptions, the RLMI documents would 
be enhanced with a flag which indicates whether the RLS is authoritative 
for the resource.

Finally, we would mention in the document that future work may define a 
mechanism by which the subscriber can pass the RLS an authorization 
token; the RLS could include this token in requests to show that it has 
been authorized by the user to send SUBSCRIBE messages for arbitrary 
resources on the user's behalf (mostly likely for a limited time period).

Of the solutions we've seen so far, I think this one is the cleanest 
from a security perspective. Further, it will almost certainly allow us 
to progress the draft now, and separate the delegation problem so that 
it can be solved in a way that works well with the other identity and 
authentication work currently underway (and might even be reusable in 
other applications, such as dial-out conference bridges).

Thoughts?

/a

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Dear All,

CALL FOR ONE ENUM INDEPENDENT EXPERT AND VOLUNTEERS TO CONTRIBUTE TO THE =
TASK FORCE TO SUPPORT THE 1ST ENUM INTEROPERABILITY EVENT (Spring 2005)

The ETSI Plugtests Interoperability service, hosted the first ENUM =
Workshop event in February 2004 to serve the whole global market in =
cooperation with the ETSI TC TISPAN committee. The ETSI Plugtests =
service is independent and has, over the last 4 years, gained a unique =
experience in organising 50 events for a broad range of domains. Further =
information on the service is available at www.etsi.org/plugtests. =20

While being part of the ETSI structure, this service is self-financed =
and fully independent. Half of the organized events are related to non =
ETSI Standards (e.g. IPv6, SIP, Bluetooth, etc).

The ETSI Plugtests service is now preparing a 1st ENUM interoperability =
event to be held spring 2005 with the objective of having a broad =
technical scope to address ENUM technologies.
Preliminary ENUM tests could be conducted during the 1st NGN Plugtests =
from 29-Nov to 3-Dec:
http://www.etsi.org/plugtests/NGN_VoIP.htm =
<http://www.etsi.org/plugtests/NGN_VoIP.htm>=20

To serve all the global ENUM solutions and all involved companies in an =
open, independent and transparent process, ETSI is looking for further =
assistance and support in two directions:

1)       CALL FOR AN INDEPENDENT EXPERT

ETSI Plugtests Service is looking for appointing an independent expert =
mainly to assist in the definition of the ENUM interoperability event, =
its architecture, its test bed and on the test cases to be performed. =
This expert will be under contractual arrangement with ETSI and under =
the financial rules of the EU eEurope programme. He will coordinate the =
work under the supervision of both the ETSI TC TISPAN committee and an =
independent task force to be set-up for providing main input to the =
programme. The expert effort can be estimated in a range of 40 to 80 =
days.

Should you like to be candidate for such expertise, please send your =
application and references to philippe.cousin@etsi.org (CC =
patrick.guillemin@etsi.org and plugtests@etsi.org) before 15 November =
2004.

2)       CALL FOR ADDITIONNAL VOLUNTEERS TO CONTRIBUTE TO A FIRST ENUM  =
INTEROPERABILITY EVENT  TASK FORCE

ETSI Interoperability Service is looking for volunteers to extend the =
task force (EPTF, ENUM Plugtests Task Force) for providing inputs and =
guidance to the 1st Interoperability event in a more open, transparent =
and independent manner as possible . The main objective of this task =
force assisted by dedicated resources of an independent expert is to =
provide guidance of the architecture, the test beds, and the test cases. =
This 1st interoperability event should address all the interoperability =
concerns for the ENUM solutions.

Should you like to be candidate to participate to this Task force, =
please send your application and references to =
Patrick.guillemin@etsi.org CC plugtests@etsi.org before 15 November =
2004.  Any expert is welcome, even non-ETSI members as long as the =
expert is committed to actively and constructively contribute to the =
work. It is expected that most of the work could be done by email and =
audio-conference but meeting(s) can also be convened when necessary.

I am looking forward to getting a lot of applications and support

Best regards

Patrick GUILLEMIN, Coordinator of EPTF
ETSI - Plugtests Technical Coordinator
http://www.etsi.org/plugtests
tel +33 (0)4 92 94 43 31

and
Philippe COUSIN
Interoperability Service Manager
+ 33 (0)4 92 94 43 06

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Communications Resource Priority for the Session 
			  Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Polk
	Filename	: draft-ietf-sip-resource-priority-05.txt
	Pages		: 33
	Date		: 2004-10-26
	
This document defines two new SIP header fields for communicating 
   resource priority, namely "Resource-Priority" and "Accept-Resource-
   Priority".  The "Resource-Priority" header field can influence the 
   behavior of SIP user agents, such as telephone gateways and IP 
   telephones, and Session Initiation Protocol (SIP) proxies.  It does 
   not directly influence the forwarding behavior of IP routers.

A URL for this Internet-Draft is:
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From sip-bounces@ietf.org  Tue Oct 26 20:22:21 2004
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Date: Tue, 26 Oct 2004 17:31:27 -0400
From: Mauricio Arango <Mauricio.Arango@Sun.COM>
Subject: Re: [Sip] verifying identity of sender
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Use of SAML in SIP doesn't require federation, nor
does it require Web Services, and pre-associations are minimal.
Additional explanations as follows:

- A Web Services stack isn't required when SIP is used to
transport  SAML assertions (or references to it). SAML
comprises assertions and protocols. Use of SAML
for the scenarios we are discusing would not involve use of
SAML protocols.

- Use of SAML to verify the 'From' identity in SIP
doesn't require federation. Federation is the linking of
one identity in one administrative domain with another
identity in another administrative domain.  Verification
of the 'From' identity doesn't demand its linking with
another identity at the destination of the SIP request.
Federation isn't a mandatory step for any use of SAML
assertions.

- Pre-associations for the scenarios under consideration
are minimal. A SAML assertion would refer to
its governing schema in the usual XML/namespaces fashion,
which would either require pre-configuration of the schema
or on-the-fly loading. Another aspect that requires
pre-association is agreement of the method to handle
a signed assertion received from the authentication proxy.
This part would have to be handled independently of whether
SAML is used or not.

- With respect to support for XML.
XML parsers could be an issue in limited-capability
user devices, though there is a increasing number
of XML parsers for small devices. Nevertheless,
XML processing for SIP identity would only be
required by the authentication proxy at the
requesting side and by the receiving side domain
proxy, which are usually servers. Also, there is
precendent of using XML in SIP as in event list
notifications.

Regards,

Mauricio Arango


"Peterson, Jon" wrote:

> > -----Original Message-----
> > From: Mauricio Arango [mailto:Mauricio.Arango@Sun.COM]
> >
> > For clarification, is the comment of "SAML heavy" because of the
> > pre-associations only?
>
> Pre-association and federation, agreement on various SAML attribute and so
> on is one aspect of it. I also think the support for XML, web services, and
> so on cannot be universally assumed in a SIP environment - a widespread
> mandate for SAML support would entail a lot of additional development,
> beyond SIP's current implementation challenge.
>
> Our emphasis in identity design has been on providing a very light solution,
> something that doesn't require us to diverge significantly from the
> operational models and implementation already described in the core
> standard. We also want to provide something for identity that is universal,
> and useful in essentially all contexts that SIP is used. I don't think we
> can position SAML to play that part in SIP's architecture, though it can
> certainly provide valuable supplemental security assertions for some
> deployments.
>
> Jon Peterson
> NeuStar, Inc.
>
> >
> > Sounds like a good initial approach.
> >
> > Mauricio Arango
> >
> >
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Tue Oct 26 20:35:17 2004
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Subject: [Sip] Agenda information and topics poll
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As usual, we've requested two sessions at IEF 61 for SIP, and will be 
trying to focus on the high-urgency active topics and limiting 
discussion on the lower-urgency (by rough consensus and impinging 
liaison statement) topics. As usual, I expect to be downright mean about 
saying "no" and getting enough time to actually make SOME progress on 
key topics.

I'll be drafting out an agenda and posting it on the softarmor web site 
over the next couple of days.

I've already receieved some suggestions from folks, using Gonzalo's 
nifty template format:

    <tr>
      <td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
Gonzalo Camarillo</a></td>
      <td>Exploders</td>
      <td><center>5</center></td>
      <td>
           <a
href="http://www.ietf.org/internet-drafts/draft-camarillo-sipping-exploders-02.txt"> 

draft-camarillo-sipping-exploders-02.txt
           </a>
        </td>
    </tr>

If you have additional topics that you feel need discussion, please send 
in suggestions, using Gonzalo's format if you can figure it out, or just:

    Who:
    Topic:
    Time needed:
    Draft:

if you can't.

I'll try and get those posted onto softarmor in an Agenda Request 
section as usual, and will confirm receipt of requests as I get them.

URL:

http://www.softarmor.com/sipwg

Thanks, y'all.

--
Dean


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This list is for NEW development of the core SIP Protocol
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Hi all,

Since this version should be ready for WGLC, a very detailed list of the
changes is provided in the document, annotated as to the source, so I won't
repeat those details here.   The majority of the changes were the issues
discussed at IETF-60, along with the agreements there to change the text to
non-normative in section 4 (Protocol structure) and to add some detail per
Rohan's comment on the necessary processing should TLS not be available.  In
addition, I received some proposed changes on a marked up hardcopy at
IETF-61 from Eric Burger.  

The only items not discussed at IETF-60 were 2 items that came up on the
list (one posted by John Elwell on August 18th on handling of privacy in
responses) the other on Oct. 15th around the appropriate character format
for the escaped headers in the URI. And, as always, there are various minor
editorial changes here and there while I was "in the area".  So, there
should be no surprises with any of the changes, although, it is possible
that there are still errors and nits that can be identified during WGLC.   

Thanks,
Mary


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Internet-Drafts@ietf.org
Sent: Monday, October 25, 2004 3:07 PM
To: i-d-announce@ietf.org
Cc: sip@ietf.org
Subject: [Sip] I-D ACTION:draft-ietf-sip-history-info-04.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Protocol Working Group
of the IETF.

	Title		: An Extension to the Session Initiation Protocol
for Request History Information
	Author(s)	: M. Barnes
	Filename	: draft-ietf-sip-history-info-04.txt
	Pages		: 47
	Date		: 2004-10-25
	
This draft defines a standard mechanism for capturing the history 
   information associated with a SIP request.  This capability enables 
   many enhanced services by providing the information as to how and why 
   a call arrives at a specific application or user.  This draft defines 
   a new optional SIP header, History-Info, for capturing the history 
   information in requests. A new option tag, Histinfo, to be included 
   in the Supported header, is defined to allow UAs to indicate whether 
   the History-Info should be returned in responses to a request which 
   has captured the history information. A new priv-value, history, is 
   added to the Privacy header to allow for privacy handling of the 
   History-Info header.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-04.txt

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<P><FONT SIZE=3D2>Hi all,</FONT>
</P>

<P><FONT SIZE=3D2>Since this version should be ready for WGLC, a very =
detailed list of the changes is provided in the document, annotated as =
to the source, so I won't repeat those details here.&nbsp;&nbsp; The =
majority of the changes were the issues discussed at IETF-60, along =
with the agreements there to change the text to non-normative in =
section 4 (Protocol structure) and to add some detail per Rohan's =
comment on the necessary processing should TLS not be available.&nbsp; =
In addition, I received some proposed changes on a marked up hardcopy =
at IETF-61 from Eric Burger.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>The only items not discussed at IETF-60 were 2 items =
that came up on the list (one posted by John Elwell on August 18th on =
handling of privacy in responses) the other on Oct. 15th around the =
appropriate character format for the escaped headers in the URI. And, =
as always, there are various minor editorial changes here and there =
while I was &quot;in the area&quot;.&nbsp; So, there should be no =
surprises with any of the changes, although, it is possible that there =
are still errors and nits that can be identified during =
WGLC.&nbsp;&nbsp; </FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Mary</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>] =
On Behalf Of Internet-Drafts@ietf.org</FONT>
<BR><FONT SIZE=3D2>Sent: Monday, October 25, 2004 3:07 PM</FONT>
<BR><FONT SIZE=3D2>To: i-d-announce@ietf.org</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] I-D =
ACTION:draft-ietf-sip-history-info-04.txt</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>A New Internet-Draft is available from the on-line =
Internet-Drafts directories.</FONT>
<BR><FONT SIZE=3D2>This draft is a work item of the Session Initiation =
Protocol Working Group of the IETF.</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
An Extension to the Session Initiation Protocol for Request History =
Information</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. =
Barnes</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-ietf-sip-history-info-04.txt</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
47</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2004-10-25</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>This draft defines a standard mechanism for =
capturing the history </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; information associated with a SIP =
request.&nbsp; This capability enables </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; many enhanced services by providing the =
information as to how and why </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; a call arrives at a specific =
application or user.&nbsp; This draft defines </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; a new optional SIP header, =
History-Info, for capturing the history </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; information in requests. A new option =
tag, Histinfo, to be included </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; in the Supported header, is defined to =
allow UAs to indicate whether </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; the History-Info should be returned in =
responses to a request which </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; has captured the history information. A =
new priv-value, history, is </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; added to the Privacy header to allow =
for privacy handling of the </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; History-Info header.</FONT>
</P>

<P><FONT SIZE=3D2>A URL for this Internet-Draft is:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-=
04.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-his=
tory-info-04.txt</A></FONT>
</P>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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On Oct 26, 2004, at 2:06 AM, Adam Roach wrote:

> After a discussion with Dean about some of the issues surrounding the 
> back-end credential issue that I raised earlier on the list, I struck 
> upon a solution that might get us past this logjam.
>
> The lynch pin to this solution is that the RLS does not ever pretend 
> to be the user. Instead, it always subscribes as its own identity. It 
> can present credentials as necessary to authenticate itself, but only 
> as itself, not as its subscribers.
>
This factors the security issues, but introduces an inconvenience in 
that any authorization policies have to know about the RLS in addition 
to the user on behalf of whom the RLS is doing the list management. 
Whether this constitutes a major or minor inconvenience is worth 
pondering.

> A consequence of this is that, lacking any other mechanism, the 
> resources in a resource list for which the RLS is not authoritative 
> will contain only the information that their notifier is willing to 
> reveal to the RLS -- which is generally the same information they 
> would render to unauthenticated users. In some cases, this might 
> result in no useful information being transferred.
>
This strikes me as closer to "major" than "minor" if an important use 
case is buddy lists - which I suppose it is. If the RLS is in fact some 
kind of "shared resource" and doesn't have an identity controllable by 
the user it is acting on behalf of there would be little motivation for 
anyone to trust it and hence policy devolves to the anonymous 
subscription case in most instances. That would seem to do substantial 
damage to the utility for buddy lists.

> Of course, the resource would still be present in the RLMI document; 
> this gives the subscriber the *option* to directly subscribe to any 
> resources for which the RLS is not authoritative. To facilitate the 
> client's ability to perform such subscriptions, the RLMI documents 
> would be enhanced with a flag which indicates whether the RLS is 
> authoritative for the resource.
>
Sure, but that's just saying that the RLS optimizations are of less 
value, and the subscriber's job of sorting through the policy versus 
overhead tradeoffs is made harder.

> Finally, we would mention in the document that future work may define 
> a mechanism by which the subscriber can pass the RLS an authorization 
> token; the RLS could include this token in requests to show that it 
> has been authorized by the user to send SUBSCRIBE messages for 
> arbitrary resources on the user's behalf (mostly likely for a limited 
> time period).
>
That's a form of delegation which seems like it would work well for 
this purpose.

> Of the solutions we've seen so far, I think this one is the cleanest 
> from a security perspective. Further, it will almost certainly allow 
> us to progress the draft now, and separate the delegation problem so 
> that it can be solved in a way that works well with the other identity 
> and authentication work currently underway (and might even be reusable 
> in other applications, such as dial-out conference bridges).
>
True, but are the tradeoffs with other aspects of utility and 
complexity right? I'm not sure but on the small amount of thought I've 
given this it strikes me as questionable.

> Thoughts?
>
One option is to delay RLS until we have delegation (at least in the 
limited context of RLS) solved, since as you point out above 
subscribers may have to fall back to separate subscriptions to all the 
resources anyway.

I'll observe that once a spec gets through to RFC, the enthusiasm to 
enhance it to do its full job wanes, and the complexities of multiple 
variants in the the wild reduces the practical utility of the "correct" 
solution.

I don't feel strongly about this one way or the other.

Dave.

> /a
>
> _______________________________________________
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>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com

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From sip-bounces@ietf.org  Wed Oct 27 09:22:55 2004
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Adam Roach wrote:

>    * Add new SIP header field (or maybe method) for credential upload
> 
>    One very simple solution would be to add a new header which contains
>    a triple of [realm,userid,password]. We would specify that this
>    header is disallowed except over SIPS connections. The client would
>    include one or more such headers in its SUBSCRIBE request, and the
>    RLS would use them to obtain information on the user's behalf.

Dave then espressed reservations over this because it grants too much 
capability to the RLS.

I agree with Dave's concern over this, and think his suggestion to 
tackle the problem of delegation head on is the right solution in the 
long term. OTOH, it doesn't seem like we can wait that long to get 
something going for RLS.

To pursue Adam's suggestion without giving away the ranch, maybe 
presence servers could have one set of credentials that grant only 
limited access (subscription by buddies), and another for full 
permissions. This is really just an implementation technique and so we 
couldn't standardize it, but it might be a way to allow this crippled 
mechanism to be useful.

	Paul


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From sip-bounces@ietf.org  Wed Oct 27 19:58:20 2004
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Subject: [Sip] Simultaneous SIP Transactions from one UA in the same dialog
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Greetings:

I am trying to find what SIP spec. has to say for the following cases: Say,
UserA called UserB via P1, a Record Routing Proxy; UserA and UserB are
connected. 

1) UserA sends ReINVITE, and BYE to UserB before receiving response to
ReINVITE. I understand the UAS behavior is specified in the spec. What about
the Proxy behavior?

2) UserA sends REINVITE and OPTIONS at the same time to UserB. Is it
allowed?

Thanks
Ragunath

--- Snippet from sip spec ---
15.1 Terminating a Session with a BYE Request

15.1.2 UAS Behavior

   The UAS MUST still respond to any pending requests received for that
   dialog.  It is RECOMMENDED that a 487 (Request Terminated) response
   be generated to those pending requests.
---

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Subject: RE: [Sip] Event Lists: Back-End Credentials
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]On=20
> Behalf Of ext
> Adam Roach
> Sent: 26.October.2004 09:06
> To: sip@ietf.org
> Cc: Jonathan Rosenberg
> Subject: [Sip] Event Lists: Back-End Credentials
>=20
>=20
> After a discussion with Dean about some of the issues surrounding the=20
> back-end credential issue that I raised earlier on the list, I struck=20
> upon a solution that might get us past this logjam.
>=20
> The lynch pin to this solution is that the RLS does not ever=20
> pretend to=20
> be the user. Instead, it always subscribes as its own=20
> identity. It can=20
> present credentials as necessary to authenticate itself, but only as=20
> itself, not as its subscribers.
>=20
> A consequence of this is that, lacking any other mechanism, the=20
> resources in a resource list for which the RLS is not=20
> authoritative will=20
> contain only the information that their notifier is willing=20
> to reveal to=20
> the RLS -- which is generally the same information they would=20
> render to=20
> unauthenticated users. In some cases, this might result in no useful=20
> information being transferred.

This limitation by itself kills the idea, IMO. Most presence lists will =
rely on the event list, and if its gonna mean that no useful information =
will come to the watcher, then the whole mecahnism is useless.

Why isn't it enough to say that the way an watcher passes the key to the =
RLS is outside the scope of this document? We have certainly made such =
assumptions in the past.

We can suggest HTML. We can suggest an email being sent to the =
administrator. We can even suggest an XCAP usage document, but I don't =
understand why the draft needs to define that explicitly.

/Hisham

>=20
> Of course, the resource would still be present in the RLMI document;=20
> this gives the subscriber the *option* to directly subscribe to any=20
> resources for which the RLS is not authoritative. To facilitate the=20
> client's ability to perform such subscriptions, the RLMI=20
> documents would=20
> be enhanced with a flag which indicates whether the RLS is=20
> authoritative=20
> for the resource.
>=20
> Finally, we would mention in the document that future work=20
> may define a=20
> mechanism by which the subscriber can pass the RLS an authorization=20
> token; the RLS could include this token in requests to show=20
> that it has=20
> been authorized by the user to send SUBSCRIBE messages for arbitrary=20
> resources on the user's behalf (mostly likely for a limited=20
> time period).
>=20
> Of the solutions we've seen so far, I think this one is the cleanest=20
> from a security perspective. Further, it will almost=20
> certainly allow us=20
> to progress the draft now, and separate the delegation=20
> problem so that=20
> it can be solved in a way that works well with the other identity and=20
> authentication work currently underway (and might even be reusable in=20
> other applications, such as dial-out conference bridges).
>=20
> Thoughts?
>=20
> /a
>=20
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

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hisham.khartabil@nokia.com wrote:

> Why isn't it enough to say that the way an watcher passes the key to 
> the RLS is outside the scope of this document?


Because that's exactly what the document says right now, and the IESG 
won't let it pass as a result.


/a

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To: hisham.khartabil@nokia.com
Subject: Re: [Sip] Event Lists: Back-End Credentials
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hisham.khartabil@nokia.com wrote:

>
>>A consequence of this is that, lacking any other mechanism, the resources in a resource list for which the RLS is not authoritative will contain only the information that their notifier is willing to reveal to the RLS -- which is generally the same information they would render to unauthenticated users. In some cases, this might result in no useful information being transferred.
>>    
>>
>
>This limitation by itself kills the idea, IMO. Most presence lists will rely on the event list, and if its gonna mean that no useful information will come to the watcher, then the whole mecahnism is useless.
>  
>

That's a bit of a strong characterization, and it seems to ignore the 
remainder of the mail that I sent out; specifically:

   1. Users have the ability to subscribe to any resources that the RLS
      marks itself as non-authoritative, and
   2. (More Importantly) Provisions for delegation of authority are
      included -- they're just not in-scope for this document.

/a

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Subject: RE: [Sip] Event Lists: Back-End Credentials
Date: Thu, 28 Oct 2004 17:35:25 +0300
Message-ID: <5816828233DEFA41807A6CFDFDF2343C3A8BDA@esebe056.ntc.nokia.com>
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I didn't know that communication with IESG is one way only. Can the IESG =
member who is blocking this explain why this cannot be outside the scope =
of this document?

Anyway, so how about an XCAP usage document that is carried signed and =
encrypted to the server using HTTP. That xcap usage document can include =
realm, username and password for realms that the user knows will be =
needed by the RLS.

Note that this is useful not just for backend subscriptions, but also =
for any list usage we can think for that will result in backend requests =
being sent on behalf of a client.

A server needing a secret key to use on behave of a user can look in the =
XCAP document of that user.

Regards,
Hisham

> -----Original Message-----
> From: ext Adam Roach [mailto:adam@nostrum.com]
> Sent: 28.October.2004 17:16
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Event Lists: Back-End Credentials
>=20
>=20
> hisham.khartabil@nokia.com wrote:
>=20
> > Why isn't it enough to say that the way an watcher passes=20
> the key to=20
> > the RLS is outside the scope of this document?
>=20
>=20
> Because that's exactly what the document says right now, and the IESG=20
> won't let it pass as a result.
>=20
>=20
> /a
>=20

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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Subject: Re: [Sip] Event Lists: Back-End Credentials
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On Oct 28, 2004, at 10:35 AM, hisham.khartabil@nokia.com wrote:

> I didn't know that communication with IESG is one way only. Can the 
> IESG member who is blocking this explain why this cannot be outside 
> the scope of this document?
>
> Anyway, so how about an XCAP usage document that is carried signed and 
> encrypted to the server using HTTP. That xcap usage document can 
> include realm, username and password for realms that the user knows 
> will be needed by the RLS.
>
That's precisely the problem. Now the RLS can impersonate the user and 
do anything the user could.

> Note that this is useful not just for backend subscriptions, but also 
> for any list usage we can think for that will result in backend 
> requests being sent on behalf of a client.
>
> A server needing a secret key to use on behave of a user can look in 
> the XCAP document of that user.
>
Can I be your RLS server? Please? Pretty please?

Dave.


> Regards,
> Hisham
>
>> -----Original Message-----
>> From: ext Adam Roach [mailto:adam@nostrum.com]
>> Sent: 28.October.2004 17:16
>> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Event Lists: Back-End Credentials
>>
>>
>> hisham.khartabil@nokia.com wrote:
>>
>>> Why isn't it enough to say that the way an watcher passes
>> the key to
>>> the RLS is outside the scope of this document?
>>
>>
>> Because that's exactly what the document says right now, and the IESG
>> won't let it pass as a result.
>>
>>
>> /a
>>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From sip-bounces@ietf.org  Thu Oct 28 19:07:32 2004
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Subject: [Sip] Working Group Last Call for History Info
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I believe History Info is pretty much ready for last call.

See draft-ietf-sip-history-info-04.txt

I believe it need news IPR boilerplate before moving to the IESG, but 
that shouldn't stop us from making a  last effort at reviewing the text 
here. Mary has provided a detailed report of changes under separate cover.

WGLC therefore starts immediately. Given the upcoming meeting, I would 
like to conclude WGLC on November 29, 2004.

We should expect to TALK about this in DC, so do your homework, kids!

--
Dean
co-chair, SIP



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From sip-bounces@ietf.org  Thu Oct 28 19:14:48 2004
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Some Advice about IPR bolierplate. USE THE NEW STUFF!
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I've noticed that several of the drafts submitted in "the rush" before 
IETF 61 haven't had their IPR boilerplate updated. RFC 3667 and 3668 
revised the requirements for IETF publication. Please read them.

We should (ok, MUST) all start using the new format. If you use xml2rfc 
(and you should!) the newer versions will mostly take care of this for you.

Here's an example from a draft that used xml2rfc (Thanks, Eric!). Note 
the critical pieces under "Status of this Memo":

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

A Mechanism for Content Indirection in Session Initiation Protocol (SIP) 
Messages
draft-ietf-sip-content-indirect-mech-05

Status of this Memo

This document is an Internet-Draft and is subject to all provisions
of section 3 of RFC 3667. By submitting this Internet-Draft, each
author represents that any applicable patent or other IPR claims of
which he or she is aware have been or will be disclosed, and any of
which he or she become aware will be disclosed, in accordance with
RFC 3668.

Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as
Internet-Drafts.

Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."

The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt.

The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.

This Internet-Draft will expire on April 25, 2005.

Copyright Notice

Copyright (C) The Internet Society (2004).

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

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Subject: RE: [Sip] Event Lists: Back-End Credentials
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> -----Original Message-----
> From: ext David R Oran [mailto:oran@cisco.com]
> Sent: 28.October.2004 18:17
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: adam@nostrum.com; sip@ietf.org
> Subject: Re: [Sip] Event Lists: Back-End Credentials
>=20
>=20
>=20
> On Oct 28, 2004, at 10:35 AM, hisham.khartabil@nokia.com wrote:
>=20
> > I didn't know that communication with IESG is one way only. Can the=20
> > IESG member who is blocking this explain why this cannot be outside=20
> > the scope of this document?
> >
> > Anyway, so how about an XCAP usage document that is carried=20
> signed and=20
> > encrypted to the server using HTTP. That xcap usage document can=20
> > include realm, username and password for realms that the user knows=20
> > will be needed by the RLS.
> >
> That's precisely the problem. Now the RLS can impersonate the=20
> user and=20
> do anything the user could.

I thought the problem was how to transport the secret to the RLS. =
Irrespective of how that is done (using XCAP or carrying it in the =
SUBSCRIBE request itself), you will face the same problem you describe =
above. So am I right in assuming that you're advocating Adam's latest =
suggestion on RLS doing backend subscription using its own address?

>=20
> > Note that this is useful not just for backend=20
> subscriptions, but also=20
> > for any list usage we can think for that will result in backend=20
> > requests being sent on behalf of a client.
> >
> > A server needing a secret key to use on behave of a user=20
> can look in=20
> > the XCAP document of that user.
> >
> Can I be your RLS server? Please? Pretty please?

Ok, but if you behave :) You would expect a trust relationship between =
the client and the RLS.

/Hisham

>=20
> Dave.
>=20
>=20
> > Regards,
> > Hisham
> >
> >> -----Original Message-----
> >> From: ext Adam Roach [mailto:adam@nostrum.com]
> >> Sent: 28.October.2004 17:16
> >> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> >> Cc: sip@ietf.org
> >> Subject: Re: [Sip] Event Lists: Back-End Credentials
> >>
> >>
> >> hisham.khartabil@nokia.com wrote:
> >>
> >>> Why isn't it enough to say that the way an watcher passes
> >> the key to
> >>> the RLS is outside the scope of this document?
> >>
> >>
> >> Because that's exactly what the document says right now,=20
> and the IESG
> >> won't let it pass as a result.
> >>
> >>
> >> /a
> >>
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>=20
>=20

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Subject: [Sip] Are we ready to WGLC identity-03?
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I polled for comments on draft-ietf-sip-identity-03 a while back, and I 
think the only direct comment I saw back was from MAT and basically 
asked "How are the implementations going?"

Is anybody implementing this? I'll admit personally I need something 
like this in products I work with, but I know nothing about 
implementation plans. I'd also be intersted in hearing any details about 
variants that people do have implementation experience with, such as 
MASS-like approaches, if any.

Its a bit hard to judge consensus in a vacuum. I'll admit the material 
here is a bit esoteric, but it does have some serious real-world 
implications. Somebody out there has an opinion, I'm sure of it . . .

If you think STRONGLY that we either are, or are not, ready for a 
working group last call on this document, please speak up.

Thanks,

--
Dean
as SIP co-chair


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Dean Willis wrote:

>
> I believe History Info is pretty much ready for last call.
>
> See draft-ietf-sip-history-info-04.txt
>
> I believe it need news IPR boilerplate before moving to the IESG, but 
> that shouldn't stop us from making a  last effort at reviewing the 
> text here. Mary has provided a detailed report of changes under 
> separate cover.
>
I believe I was wrong about that -- apparently, I made the "fix the 
boilerplate" mark on the wrong row of my checklist. Sorry about that! 
Mary, as usual, did it right, and I, as happens far too often, did not.

--
Dean

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Subject: Re: [Sip] Event Lists: Back-End Credentials
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On Oct 28, 2004, at 6:52 PM, <hisham.khartabil@nokia.com> wrote:

>
>
>> -----Original Message-----
>> From: ext David R Oran [mailto:oran@cisco.com]
>> Sent: 28.October.2004 18:17
>> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
>> Cc: adam@nostrum.com; sip@ietf.org
>> Subject: Re: [Sip] Event Lists: Back-End Credentials
>>
>>
>>
>> On Oct 28, 2004, at 10:35 AM, hisham.khartabil@nokia.com wrote:
>>
>>> I didn't know that communication with IESG is one way only. Can the
>>> IESG member who is blocking this explain why this cannot be outside
>>> the scope of this document?
>>>
>>> Anyway, so how about an XCAP usage document that is carried
>> signed and
>>> encrypted to the server using HTTP. That xcap usage document can
>>> include realm, username and password for realms that the user knows
>>> will be needed by the RLS.
>>>
>> That's precisely the problem. Now the RLS can impersonate the
>> user and
>> do anything the user could.
>
> I thought the problem was how to transport the secret to the RLS. 
> Irrespective of how that is done (using XCAP or carrying it in the 
> SUBSCRIBE request itself), you will face the same problem you describe 
> above. So am I right in assuming that you're advocating Adam's latest 
> suggestion on RLS doing backend subscription using its own address?
>
I'm not sure what I'm "advocating", since I said in my original post 
that I don't feel strongly about the subject. Since I'm not in the 
middle of the fray amongst the folks who think we need RLS urgently, I 
can comfortably sit back and "advocate" not progressing RLS until we 
have at least a resticted subset of the delegation problem solved.

If the only alternatives on the table are whether to progress RLS with 
a security scheme that opens the user up to impersonation by a server 
he has very limited trust in, and progressing an solution that uses 
deprate credentials and hence either has limited utility or high user 
complexity (users have to directly subscribe or be limited to 
information available to the whole world), I guess I would vote for the 
latter.

>>
>>> Note that this is useful not just for backend
>> subscriptions, but also
>>> for any list usage we can think for that will result in backend
>>> requests being sent on behalf of a client.
>>>
>>> A server needing a secret key to use on behave of a user
>> can look in
>>> the XCAP document of that user.
>>>
>> Can I be your RLS server? Please? Pretty please?
>
> Ok, but if you behave :) You would expect a trust relationship between 
> the client and the RLS.
>
Yes, but a very limited one, not one that permits identity theft.

> /Hisham
>
>>
>> Dave.
>>
>>
>>> Regards,
>>> Hisham
>>>
>>>> -----Original Message-----
>>>> From: ext Adam Roach [mailto:adam@nostrum.com]
>>>> Sent: 28.October.2004 17:16
>>>> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
>>>> Cc: sip@ietf.org
>>>> Subject: Re: [Sip] Event Lists: Back-End Credentials
>>>>
>>>>
>>>> hisham.khartabil@nokia.com wrote:
>>>>
>>>>> Why isn't it enough to say that the way an watcher passes
>>>> the key to
>>>>> the RLS is outside the scope of this document?
>>>>
>>>>
>>>> Because that's exactly what the document says right now,
>> and the IESG
>>>> won't let it pass as a result.
>>>>
>>>>
>>>> /a
>>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>> David R. Oran
>> Cisco Fellow
>> Cisco Systems
>> 7 Ladyslipper Lane
>> Acton, MA 01720 USA
>> Tel: +1 978 264 2048
>> Email: oran@cisco.com
>>
>>
>>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com

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Subject: RE: [Sip] Call Transfer Scenario
Date: Wed, 20 Oct 2004 08:53:50 -0500
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Thread-Topic: [Sip] Call Transfer Scenario
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From: "Bai, Gali" <gbai@ti.com>
To: "Anurag Kabra" <anuragk@aftek.com>, <sip@ietf.org>
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Anurag,

In the attended call transfer scenario, A-transferor needs to call
C-transfer target first and if C agrees to accept call from B, A will
send REFER with Replaces in its Refer-to header to B-transferee; And the
Invite from B to C also contains Replaces header to C.

So the differences are:
1. In attended call transfer, A calls C first and sends REFER with
Replaces in the Refer-to header; Unattended call transfer, A does not
need to call C, just sends REFER to B;
2. B Invite C, in attended call transfer, the Invite contains Replaces
header to replace the dialog between A and C; in unattended call
transfer, B Invite to C does not have Replaces header
3. If C receives Invite with Replaces header, it will be answered as an
attended call transfer, C needs to replace the existing dialog based on
Replaces header.=20


Gali Bai

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Anurag Kabra
Sent: Tuesday, October 19, 2004 12:18 PM
To: sip@ietf.org
Subject: [Sip] Call Transfer Scenario

Hi

     I need to clear few of my doubts regarding call transfer,=20

In a scenario of A, B and C - Unattended Transfer (Blind Transfer)

A =3D transferor
B =3D transferee
C =3D transfer target

1. There is a call set up between A and B
2.  A need to transfer to C - A puts the RTP session on Hold with
Re-INVITE=20
and sends REFER with refer-to C,=20
3. B on getting refer needs to put audio streaming on hold and so sends

Re-INVITE!! -- (DOUBT - is this step right)
4. B sends NOTIFY with subscription-state =3D active and expires to A
5. B now INVITES C=20
6. On getting response its sends a NOTIFY to A with subscription-state =
=3D

terminated
(Need to know if the REFER expires because of expires in NOTIFY what can
be=20
done ie if we get a time out on A side)
7. Now A sends a BYE to B
(What to do if B wants to terminate the dialog with CANCEL or BYE)

How can we differentiate Attended Transfer and Blind Transfer in the=20
application??

Can anyone tell me free sip UA implementing Call Transfer?

Please do let me know any of your views. Please correct me if i am wrong

anywhere.=20

Best Regards
  Anurag.

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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----------tnhljmsuzxutycwonqbd
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Thread-Topic: Lack of QoS reservation resonsibility negotiation in RFC3312?
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From: "Xin Chen" <Xin.Chen@uk.fujitsu.com>
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Subject: [Sip] Lack of QoS reservation resonsibility negotiation in RFC3312?
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Hi,=20

I have a question regarding RFC3312 "Integration of Resource Management
and Session Initiation Protocol (SIP)".

This specification seems only specify the negotiation of desired QoS
condition, e.g. e2e. However, I think it is lack of QoS reservation
responsibility negotiation.

Some QoS reservation protocol unlike RSVP, e.g. NSIS allow user agent to
reserve QoS for bi-directions, so if both SIP UAC and UAS support such
kind of QoS reservation protocol, for a SIP session with precondition of
e2e QoS, both UAC and UAS may attempt to reserve QoS for both
directions, then double reservations for the same media flow may occur,
and I think this is not a good thing and should be avoided.

So I think during SIP session setup, the responsibility of QoS
reservation (which side reserve with direction) shall also be
negotionated.

How do you think so?

Regards
=20
Xin Chen
Mobile Network Division
Fujitsu Laboratories Europe
Tel: +44(0)2086064453
FJ coins: 795414453


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Date: Fri, 29 Oct 2004 10:31:33 -0700
Subject: Re: [Sip] Are we ready to WGLC identity-03?
From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>, "sip@ietf.org" <sip@ietf.org>
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There is open source code in www.resiprocate.org to compute the identity
hash. 


On 10/28/04 3:58 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> 
> I polled for comments on draft-ietf-sip-identity-03 a while back, and I
> think the only direct comment I saw back was from MAT and basically
> asked "How are the implementations going?"
> 
> Is anybody implementing this? I'll admit personally I need something
> like this in products I work with, but I know nothing about
> implementation plans. I'd also be intersted in hearing any details about
> variants that people do have implementation experience with, such as
> MASS-like approaches, if any.
> 
> Its a bit hard to judge consensus in a vacuum. I'll admit the material
> here is a bit esoteric, but it does have some serious real-world
> implications. Somebody out there has an opinion, I'm sure of it . . .
> 
> If you think STRONGLY that we either are, or are not, ready for a
> working group last call on this document, please speak up.
> 
> Thanks,
> 
> --
> Dean
> as SIP co-chair
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Use sipping@ietf.org for new developments on the application of sip


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Subject: [Sip] LC on nit-actions and nit-problems
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Folks -

WGLC for the nit actions and problems draft started Oct 20 and is 
scheduled to
end November 3 (next Wednesday).

So far I have received _no_ comments.

I believe that's because these drafts have been thoroughly discussed, 
are well
understood, and they are ready to go (the points where more discussion 
is needed
have all been moved to nit-future).

If you agree, please take a moment to send a note to the list saying "I 
read this draft
and believe it is ready to be published".

If you disagree - now is the time to comment!

Thanks,
RjS



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I favor a WGLC of this draft sooner, rather than later.

RjS

On Oct 28, 2004, at 5:58 PM, Dean Willis wrote:

>
> I polled for comments on draft-ietf-sip-identity-03 a while back, and 
> I think the only direct comment I saw back was from MAT and basically 
> asked "How are the implementations going?"
>
> Is anybody implementing this? I'll admit personally I need something 
> like this in products I work with, but I know nothing about 
> implementation plans. I'd also be intersted in hearing any details 
> about variants that people do have implementation experience with, 
> such as MASS-like approaches, if any.
>
> Its a bit hard to judge consensus in a vacuum. I'll admit the material 
> here is a bit esoteric, but it does have some serious real-world 
> implications. Somebody out there has an opinion, I'm sure of it . . .
>
> If you think STRONGLY that we either are, or are not, ready for a 
> working group last call on this document, please speak up.
>
> Thanks,
>
> --
> Dean
> as SIP co-chair
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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--===============1872177677==--




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Communications Resource Priority for the Session 
			  Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Polk
	Filename	: draft-ietf-sip-resource-priority-05.txt
	Pages		: 33
	Date		: 2004-10-26
	
This document defines two new SIP header fields for communicating 
   resource priority, namely "Resource-Priority" and "Accept-Resource-
   Priority".  The "Resource-Priority" header field can influence the 
   behavior of SIP user agents, such as telephone gateways and IP 
   telephones, and Session Initiation Protocol (SIP) proxies.  It does 
   not directly influence the forwarding behavior of IP routers.

A URL for this Internet-Draft is:
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From sip-bounces@ietf.org  Sun Oct 31 08:46:25 2004
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Date: Sun, 31 Oct 2004 14:33:56 +0100 (CET)
From: =?ISO-8859-1?Q?Egil_C=2E_=D8sthus?= <egilconr@stud.ntnu.no>
To: sip@ietf.org
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Subject: [Sip] SIP Extensions
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Hi,

I have a question about section 8.1.1.9 in rfc3261.
It is said in this section that "The option tags listed MUST only refer to
extensions defined in standards-track RFCs". Does this mean that the only
legal extensions to SIP is those how are described in RFCs? If I would
like to define a method in SIP, is it possible to use this method some way
and at the same time be 100% SIP compliant?

Cheers,
Egil

    =B4''`.
   : :' :
   `. `'
     `_


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


