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From: "Chris Whitaker" <whitakercd@hotmail.com>
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Subject: [Sip] Question regarding SDP and holds
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In rfc3264, it states:

If the stream to be placed on hold was previously a sendrecv media
   stream, it is placed on hold by marking it as sendonly.

So my question is this.
If the stream being altered was previously a sendrecv media stream, and has 
become a sendonly stream... how does one communicate that without implying 
that the stream is on hold [ie neither sending nor receiving]?  To be clear 
within the existing recommended proceedures for communcating a hold, I am 
trying to understand a means of differentiating when the source of media 
stream has stopped receiving but NOT stopped sending and when it has stopped 
BOTH.

Secondly, if a sendrecv stream is becoming neither a sending nor receiving 
stream [which often occurs when it goes on hold] why does the rfc suggest 
claiming it is sendonly... would not inactive be more accurate?

Chris

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Orit Levin <oritl@microsoft.com>
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Subject: [Sip] Re: GRUU, device and un-REGISTER with *
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Changed to sip, since I dont think this is a SIMPLE issue.

The meaning of Contact: * with Expires:0 remains unchanged; delete all 
contacts.

The change you are proposing would necessitate a Require in 
registrations, whereas the current gruu stuff is all backwards 
compatible and thus Supported is sufficient. I am disinclined to make 
such a change; we're not redefining the REGISTER operation as opposed to 
adding gruu, and I am unsure what real problem is being solved by such a 
change.

Thanks,
Jonathan R.

Orit Levin wrote:

> Guys,
> Now when we have GRUU, device identifier and the expanded registration 
> procedure, what would be the meaning of un-REGISTER with *?
> It makes a lot of sense to un-REGISTER a specific device with all its 
> contacts at once. Some would say that it is more useful (and reasonable 
> from authentication/security point of view) than un-REGISTERing all the 
> devices of the same user at once.
>  
> I personally think that we need both semantics and the syntax for each 
> needs to be defined/agreed upon.
>  
> Thanks,
> Orit.

-- 
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 exim@www1.ietf.org  Fri Apr  2 16:12:16 2004
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Subject: RE: [Sip] Keepalive on media protocols while mute / on-hold etc, & SIP INVITE
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comments inline
=20

> A "NAT-friendly" RTP implementation could periodically send=20
> no-ops or silence (depending on the payload types already=20
> negotiated) if it is not sending anything else - or is this=20
> outlawed by being set to "recvonly" or "inactive"?


Do we really have to use special no-ops or silence RTP packets? Why not
just send zero length payload UDP packets, then we do not have to invent
something new. An RTP stack should already today handle this, otherwise
that would be a simple attack method. As long as the UDP packets pass
through the NATs and firewalls it does not matter whether the receiver
is implemented to understand the zero length packet or not.

/ Christian Jansson, Hotsip

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Subject: [SIP] Sending  2xx responses to INVITE
Date: Mon, 5 Apr 2004 14:34:15 +0400
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Dear SIP experts,

I have following questions (maybe already discussed on the list in the past)
regarding the sending of 2xx responses to INVITE.

In what way does TU send retransmissions of 2xx responses to INVITE?
What is the interval between these retransmissions? How many retansmissions
can be sent?

Best regards
Anton


_______________________________________________
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: Bilajbegovic Damir <damir.bilajbegovic@siemens.com>
To: ??????????? ????? ?????????? <galactionov@loniis.ru>, sip@ietf.org
Subject: RE: [SIP] Sending  2xx responses to INVITE
Date: Mon, 5 Apr 2004 12:48:01 +0200 
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Hi,	
	Not sure but I think 1st Invite and 7 retransmission. Intervals are
increasing by double. 
		Hope I helped.
			Bilaj

-----Original Message-----
From: galactionov@loniis.ru [mailto:galactionov@loniis.ru] 
Sent: Monday, April 05, 2004 12:34 PM
To: sip@ietf.org
Subject: [SIP] Sending 2xx responses to INVITE


Dear SIP experts,

I have following questions (maybe already discussed on the list in the past)
regarding the sending of 2xx responses to INVITE.

In what way does TU send retransmissions of 2xx responses to INVITE? What is
the interval between these retransmissions? How many retansmissions can be
sent?

Best regards
Anton


_______________________________________________
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: "Umesh Sharma" <umesh_sharma@persistent.co.in>
To: "Bilajbegovic Damir" <damir.bilajbegovic@siemens.com>,
        "??????????? ????? ??????????" <galactionov@loniis.ru>, <sip@ietf.org>
Subject: RE: [SIP] Sending  2xx responses to INVITE
Date: Mon, 5 Apr 2004 17:35:52 +0530
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Hi

>>In what way does TU send retransmissions of 2xx responses to INVITE? What
is
>>the interval between these retransmissions?

To quote from rfc 3261 directly:

"The 2xx response is passed to the transport with an
 interval that starts at T1 seconds and doubles for each
 retransmission until it reaches T2 seconds (T1 and T2 are defined in
 Section 17).  Response retransmissions cease when an ACK request for
 the response is received.  This is independent of whatever transport
 protocols are used to send the response."

>>How many retansmissions can be sent?

"If the server retransmits the 2xx response for 64*T1 seconds without
 receiving an ACK, the dialog is confirmed, but the session SHOULD be
 terminated."

In 17.1.1.1

"Eventually, the server transaction decides
   to send a final response.  For unreliable transports, that response
   is retransmitted periodically, and for reliable transports, it is
   sent once."

I have a question too here, Now this line above seems to contradict this
paragraph in
13.3.1.4.

      "Since 2xx is retransmitted end-to-end, there may be hops between
      UAS and UAC that are UDP.  To ensure reliable delivery across
      these hops, the response is retransmitted periodically even if the
      transport at the UAS is reliable."

Am I reading this incorrectly?

regards
-umesh
-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
Bilajbegovic Damir
Sent: Monday, April 05, 2004 4:18 PM
To: ??????????? ????? ??????????; sip@ietf.org
Subject: RE: [SIP] Sending 2xx responses to INVITE


Hi,
	Not sure but I think 1st Invite and 7 retransmission. Intervals are
increasing by double.
		Hope I helped.
			Bilaj

-----Original Message-----
From: galactionov@loniis.ru [mailto:galactionov@loniis.ru]
Sent: Monday, April 05, 2004 12:34 PM
To: sip@ietf.org
Subject: [SIP] Sending 2xx responses to INVITE


Dear SIP experts,

I have following questions (maybe already discussed on the list in the past)
regarding the sending of 2xx responses to INVITE.

In what way does TU send retransmissions of 2xx responses to INVITE? What is
the interval between these retransmissions? How many retansmissions can be
sent?

Best regards
Anton


_______________________________________________
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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Message-ID: <31F31018FE2DC941A70BD30477510B0C19D643@biscm02>
From: "Hu Yijun,BISC R&D SA(BJ)" <yijun.hu@siemens.com>
To: "'Umesh Sharma'" <umesh_sharma@persistent.co.in>,
        Bilajbegovic Damir
	 <damir.bilajbegovic@siemens.com>,
        ??????????? ????? ??????????
	 <galactionov@loniis.ru>, sip@ietf.org
Subject: RE: [SIP] Sending  2xx responses to INVITE
Date: Tue, 6 Apr 2004 09:01:39 +0800 
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Hello,

Please see my comments below.

regards
hu yijun

>-----Original Message-----
>From: Umesh Sharma [mailto:umesh_sharma@persistent.co.in]
>Sent: Monday, April 05, 2004 8:06 PM
>To: Bilajbegovic Damir; ??????????? ????? ??????????; sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>Hi
>
>>>In what way does TU send retransmissions of 2xx responses to 
>INVITE? What
>is
>>>the interval between these retransmissions?
>
>To quote from rfc 3261 directly:
>
>"The 2xx response is passed to the transport with an
> interval that starts at T1 seconds and doubles for each
> retransmission until it reaches T2 seconds (T1 and T2 are defined in
> Section 17).  Response retransmissions cease when an ACK request for
> the response is received.  This is independent of whatever transport
> protocols are used to send the response."
>
>>>How many retansmissions can be sent?
>
>"If the server retransmits the 2xx response for 64*T1 seconds without
> receiving an ACK, the dialog is confirmed, but the session SHOULD be
> terminated."
>
>In 17.1.1.1
>
>"Eventually, the server transaction decides
>   to send a final response.  For unreliable transports, that response
>   is retransmitted periodically, and for reliable transports, it is
>   sent once."
>
>I have a question too here, Now this line above seems to 
>contradict this
>paragraph in
>13.3.1.4.
>
>      "Since 2xx is retransmitted end-to-end, there may be hops between
>      UAS and UAC that are UDP.  To ensure reliable delivery across
>      these hops, the response is retransmitted periodically 
>even if the
>      transport at the UAS is reliable."
>
>Am I reading this incorrectly?

The description of ch 17.1.1.1 was foucs on the retransmission of the non-2xx final response to INVITE, which were administered by the transaction layer. Those responses are delivered hop-by-hop. So that, for a certain hop, which uses reliable transport, the non-2xx final response is sent only once.

However, the retransmission of the 2XX to INVITE was administered by the transaction user (TU). It MUST comply with the ch13.3.1.4 of rfc3261: "the response is retransmitted periodically even if the transport at the UAS is reliable".

>
>regards
>-umesh
>-----Original Message-----
>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
>Bilajbegovic Damir
>Sent: Monday, April 05, 2004 4:18 PM
>To: ??????????? ????? ??????????; sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>Hi,
>	Not sure but I think 1st Invite and 7 retransmission. 
>Intervals are
>increasing by double.
>		Hope I helped.
>			Bilaj
>
>-----Original Message-----
>From: galactionov@loniis.ru [mailto:galactionov@loniis.ru]
>Sent: Monday, April 05, 2004 12:34 PM
>To: sip@ietf.org
>Subject: [SIP] Sending 2xx responses to INVITE
>
>
>Dear SIP experts,
>
>I have following questions (maybe already discussed on the 
>list in the past)
>regarding the sending of 2xx responses to INVITE.
>
>In what way does TU send retransmissions of 2xx responses to 
>INVITE? What is
>the interval between these retransmissions? How many 
>retansmissions can be
>sent?
>
>Best regards
>Anton
>
>
>_______________________________________________
>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 exim@www1.ietf.org  Tue Apr  6 07:13:54 2004
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Date: Tue, 06 Apr 2004 11:30:09 +0100
From: Alex Bligh <alex@alex.org.uk>
Reply-To: Alex Bligh <alex@alex.org.uk>
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Subject: [Sip] RFC3261 & authenticating PROXIES to other proxies & UAS's
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First, apologies if this is a dumb suggestion - it seems so
obvious that it must either be an even more obvious reason
not to do it, or someone must have thought of it before.

The (general) problem I am trying to solve looks like this:


   UAC <-----/\/\/\/\/---> P1 <--/\/--> P2 <---> UAS => PSTN
  (CPE)                 (proxy)      (proxy)  (media
                                              gateway)

P2 may or may not be present.

Let us assume UAC and P1 are very distant, and P1 & either
P2 or UAS are sufficiently distant that host security is
impractical without adding some additional layer (ipsec,
tls etc.).

UAC is CPE on a site of a customer of SP1, who operates P1.
UAS is a media gateway, and P2 a (possibly) intervening
proxy, operated by SP2.

Let us further assume that UAC registers to P1, and
that P1 has already authenticated UAC, and that any
INVITE passed through P1 has already been authenticated
as coming from UAC and been authorized by P1 according
to SP1's local policy, prior to being forwarded to P2.

Let us further assume that it is the local policy of SP1
that any such authorized calls it passes to SP2 would
be authorized by SP1. So for instance SP2 is a VoIP
termination provider, and SP1 a VoIP service provider.

SP2 wishes to authenticate & authorize calls made to UAS.
It uses HTTP Digest Authentication (per RFC3261 s.22).
This could either be done at P2, or at UAS.

RFC3261 section 16.7 suggests that Proxies MUST forward such
final 4xx responses received back towards the UAS (as there
will still be Via: header(s)).

This means that credentials for P2 or UAS must be supplied
by UAC, not by P1.

This has a number of disadvantages, for instance
1. The 401/407 response is forwarded hop-by-hop all the
   way to UAC, and a retry made. This causes unnecessary
   network traffic and delay.
2. Either
   a) UAS/P2 must challenge UAC with the same realm as P1
      did, in which case in practice SP2 must be able to
      query SP1's authentication database, which necessitates
      yet another protocol; or
   b) UAS/P2 challenges UAC in a different realm to P1, which
      means UAC must hold two sets of credentials (and, in the
      case where SP3, SP4, etc. are providing similar services
      to SP1, a possibly unlimited set of credentials).
3. In either case, UAC has credentials which allow it to traverse
   P2 / UAS, which may be undesirable from both SP1 and SP2's
   point of view, if (say) the contractual relationship is
   between User and SP1, and SP1 and SP2, but not between
   User and SP2.

What I am trying to determine is why a 401/407 challenge
from P2/UAS could not result in a response by P1, i.e. P1
itself provide the credentials to P2/UAS. Whilst this would
seem to be in breach of RC3261, I can't see what harm it
does, and it would bring considerable benefit - for instance
P1 could authenticate to P2/UAC using its own credentials,
as opposed to credentials belonging to the particular user
making the call. IE P1 detects that the response comes from
P2 with a realm that it knows, and behaves (in essence) a little
like a UAC.

So the traffic flow would look like this (assuming P1 authenticates
to P2)

 UAC              P1                 P2                UAS
  |                |                  |                 |
  |  INVITE        |                  |                 |
  | ------------>  |                  |                 |
  |                |                  |                 |
  |  407 Unauth    |                  |                 |
  | <-----------   |                  |                 |
  |                |                  |                 |
  |    ACK         |                  |                 |
  | ----------->   |                  |                 |
  |                |                  |                 |
  |  INVITE with   |                  |                 |
  |  user creds    |                  |                 |
  |  for SP1       |                  |                 |
  | ----------->   |                  |                 |
  |                |                  |                 |
  | 100 Trying     |    INVITE        |                 |
  | <-----------   |  ------------>   |                 | }
  |                |                  |                 | }
  |                |   407 Unauth     |                 | }
  |                |  <------------   |                 | }
  |                |                  |                 | }
  |                |     ACK          |                 | } new
  |                |  ------------->  |                 | }
  |                |                  |                 | }
  |                |  INVITE with     |                 | }
  |                |  SP1 creds for   |                 | }
  |                |  SP2             |                 | }
  |                |  ------------->  |                 | }
  |                |                  |                 |
  |                |   100 Trying     |    INVITE       |
  |    100 Trying  |  <-----------    | ------------->  |
  | <------------  |                  |                 |
  |                |                  |    200 OK       |
  |                |   200 OK         | <-------------  |
  |   200 OK       |  <-------------  |                 |
  | <------------- |                  |                 |
  |                |                  |                 |
  |                |                  |                 |
  |                |                  |                 |
  |                |                  |                 |

It is relatively easy to imagine variations where P1 alternatively
or additionally authenticates to UAS, or P2 alternatively or
additionally authenticates to UAS.

Now, it seems to me, apart from a slight alteration to section 16.7
of RFC3261, this could be achieved with no protocol changes. To
support it would only require a change at those proxies wishing
to authenticate themselves to other proxies or UAS's, and no change
is required at proxies or UAS's being authenticated to.

So:
1. Is this in fact not in breach of RFC3261 at all?
2. If not, is this a good solution to the problem?
3. If not, why not, and what should I do instead?


Alex

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From: Viktor.Bekesi@nokia.com
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Subject: [Sip] Caller-Preferences: Representing negated Boolean feature tags
Date: Tue, 6 Apr 2004 13:31:24 +0200
Message-ID: <DF3C159C4F4BCE4BB35B43F4AB6D3D84074919@buebe002.europe.nokia.com>
thread-topic: [Sip] Caller-Preferences: Representing negated Boolean feature tags
thread-index: AcQbyrEYPgB2CpcHRb+gH9kTJKJfzg==
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Hello,

I am implementing SIP Caller-Preferences related functionalities, and =
there is one thing that is not clear for me after reading the relevant =
drafts and rfcs: Does it make sense to negate a Boolean type feature =
tag? like the audio in the example:

   (& (! (audio=3DTRUE))
      (video=3DTRUE)
      (sip.mobility=3Dfixed)
      (message=3DTRUE)
      (| (sip.methods=3DINVITE) (sip.methods=3DOPTIONS) =
(sip.methods=3DBYE)
         (sip.methods=3DCANCEL) (sip.methods=3DACK))
      (| (sip.schemes=3Dsip) (sip.schemes=3Dhttp)))


If yes, then the next question arises: How can I represent negated =
Boolean type feature tags in SIP header parameters?=20
According to draft-ietf-sip-callerprefs-10,=20

   (& (sip.mobility=3Dfixed)
      (| (! (sip.events=3Dpresence)) (sip.events=3Dwinfo))
      (| (language=3Den) (language=3Dde))
      (sip.description=3D"PC")
      (sip.newparam=3DTRUE)
      (rangeparam=3D-4..5125/1000))

this declaration can be turned into this representation:

   =
Accept-Contact:*;mobility=3D"fixed";events=3D"!presence,winfo";language=3D=
"en,de"
    ;description=3D"<PC>";+sip.newparam;+rangeparam=3D"#-4:+5.125"

(I reversed the order as it was in the document)

Here the events are negated and that is indicated by the exclamation =
mark at the beginning of the event-tag-value. Boolean type tags are =
turned into flag parameters with no value. So then how can the very =
first example represented?


Thanks in advance, best regards:=20

Viktor Bekesi
Nokia Networks
+36-20 9849-202
=20

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From: "Umesh Sharma" <umesh_sharma@persistent.co.in>
To: "Hu Yijun,BISC R&D SA\(BJ\)" <yijun.hu@siemens.com>, <sip@ietf.org>
Subject: RE: [SIP] Sending  2xx responses to INVITE
Date: Tue, 6 Apr 2004 18:09:54 +0530
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hi Hu

I re-read the section 17.1.1.1, it looks to me a general statement which
does not
distinguishes between non-2xx final & 2xx response.

-regards
umesh

-----Original Message-----
From: Hu Yijun,BISC R&D SA(BJ) [mailto:yijun.hu@siemens.com]
Sent: Tuesday, April 06, 2004 6:32 AM
To: 'Umesh Sharma'; Bilajbegovic Damir; ??????????? ????? ??????????;
sip@ietf.org
Subject: RE: [SIP] Sending 2xx responses to INVITE


Hello,

Please see my comments below.

regards
hu yijun

>-----Original Message-----
>From: Umesh Sharma [mailto:umesh_sharma@persistent.co.in]
>Sent: Monday, April 05, 2004 8:06 PM
>To: Bilajbegovic Damir; ??????????? ????? ??????????; sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>Hi
>
>>>In what way does TU send retransmissions of 2xx responses to
>INVITE? What
>is
>>>the interval between these retransmissions?
>
>To quote from rfc 3261 directly:
>
>"The 2xx response is passed to the transport with an
> interval that starts at T1 seconds and doubles for each
> retransmission until it reaches T2 seconds (T1 and T2 are defined in
> Section 17).  Response retransmissions cease when an ACK request for
> the response is received.  This is independent of whatever transport
> protocols are used to send the response."
>
>>>How many retansmissions can be sent?
>
>"If the server retransmits the 2xx response for 64*T1 seconds without
> receiving an ACK, the dialog is confirmed, but the session SHOULD be
> terminated."
>
>In 17.1.1.1
>
>"Eventually, the server transaction decides
>   to send a final response.  For unreliable transports, that response
>   is retransmitted periodically, and for reliable transports, it is
>   sent once."
>
>I have a question too here, Now this line above seems to
>contradict this
>paragraph in
>13.3.1.4.
>
>      "Since 2xx is retransmitted end-to-end, there may be hops between
>      UAS and UAC that are UDP.  To ensure reliable delivery across
>      these hops, the response is retransmitted periodically
>even if the
>      transport at the UAS is reliable."
>
>Am I reading this incorrectly?

The description of ch 17.1.1.1 was foucs on the retransmission of the
non-2xx final response to INVITE, which were administered by the transaction
layer. Those responses are delivered hop-by-hop. So that, for a certain hop,
which uses reliable transport, the non-2xx final response is sent only once.

However, the retransmission of the 2XX to INVITE was administered by the
transaction user (TU). It MUST comply with the ch13.3.1.4 of rfc3261: "the
response is retransmitted periodically even if the transport at the UAS is
reliable".

>
>regards
>-umesh
>-----Original Message-----
>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
>Bilajbegovic Damir
>Sent: Monday, April 05, 2004 4:18 PM
>To: ??????????? ????? ??????????; sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>Hi,
>	Not sure but I think 1st Invite and 7 retransmission.
>Intervals are
>increasing by double.
>		Hope I helped.
>			Bilaj
>
>-----Original Message-----
>From: galactionov@loniis.ru [mailto:galactionov@loniis.ru]
>Sent: Monday, April 05, 2004 12:34 PM
>To: sip@ietf.org
>Subject: [SIP] Sending 2xx responses to INVITE
>
>
>Dear SIP experts,
>
>I have following questions (maybe already discussed on the
>list in the past)
>regarding the sending of 2xx responses to INVITE.
>
>In what way does TU send retransmissions of 2xx responses to
>INVITE? What is
>the interval between these retransmissions? How many
>retansmissions can be
>sent?
>
>Best regards
>Anton
>
>
>_______________________________________________
>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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>
>
>_______________________________________________
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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
>


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To: Alex Bligh <alex@alex.org.uk>
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Subject: Re: [Sip] RFC3261 & authenticating PROXIES to other proxies & UAS's
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Hi Alex!

Alex Bligh wrote:

> So:
> 1. Is this in fact not in breach of RFC3261 at all?
> 2. If not, is this a good solution to the problem?

I think (not sure) another problem is the CSeq - which will be increased 
with every request, and if the proxy P! inreases it on behalf of UAC1, 
then this might cause problems.

> 3. If not, why not, and what should I do instead?

SP2 can authenticate your IP address (IP address of P1) and all requests 
from your IP address will be billed on your account. To avaid spoffing, 
UDP should be disabled on P2 and you should send your requests using TCP.

Klaus


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From: Fredrik Thulin <ft@it.su.se>
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        Alex Bligh <alex@alex.org.uk>
Subject: Re: [Sip] RFC3261 & authenticating PROXIES to other proxies & UAS's
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On Tuesday 06 April 2004 15.36, Klaus Darilion wrote:
> Hi Alex!
>
> Alex Bligh wrote:
> > So:
> > 1. Is this in fact not in breach of RFC3261 at all?
> > 2. If not, is this a good solution to the problem?
>
> I think (not sure) another problem is the CSeq - which will be increased
> with every request, and if the proxy P! inreases it on behalf of UAC1,
> then this might cause problems.

Yes, this is even pointed out in RFC3261 #22.3 Proxy-to-User Authentication :

      If a proxy were to resubmit a request adding a Proxy-Authorization
      header field value, it would need to increment the CSeq in the new
      request.  However, this would cause the UAC that submitted the
      original request to discard a response from the UAS, as the CSeq
      value would be different.

> > 3. If not, why not, and what should I do instead?
>
> SP2 can authenticate your IP address (IP address of P1) and all requests
> from your IP address will be billed on your account. To avaid spoffing,
> UDP should be disabled on P2 and you should send your requests using TCP.

There is wording in section 22.3 saying proxys MUST NOT add things to 
authorization headers, and that all 407 responses MUST be forwarded 
upstreams. That is a pity, since I think otherwise the following could solve 
the problem :

P1 could, upon receiving the 407 from P2, start a new transaction with a 
different branch parameter and added authentication credentials. The request 
could then be re-submitted to P2.

Perhaps someone can explain what was the reason for not permitting proxy to 
proxy authentication? For example, I would have wanted to do that when 
someone chooses to redirect incoming calls to their cellular phone. In that 
case, it is not fair to send a 3xx to the caller and let them pay for the 
PSTN costs. Instead, I would like for the proxy making the redirection 
decision to send the request to my PSTN gateway, providing credentials for 
the user who requested the redirection.

/Fredrik



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--On 06 April 2004 18:39 +0200 Fredrik Thulin <ft@it.su.se> wrote:

> P1 could, upon receiving the 407 from P2, start a new transaction with a
> different branch parameter and added authentication credentials. The
> request  could then be re-submitted to P2.

Far neater. And still no need to change UAC, P2 or UAS.

Alex


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From: Alex Bligh <alex@alex.org.uk>
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To: Klaus Darilion <klaus.mailinglists@pernau.at>
Cc: sip@ietf.org, Alex Bligh <alex@alex.org.uk>
Subject: Re: [Sip] RFC3261 & authenticating PROXIES to other proxies & UAS's
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--On 06 April 2004 15:36 +0200 Klaus Darilion 
<klaus.mailinglists@pernau.at> wrote:

>
>> So:
>> 1. Is this in fact not in breach of RFC3261 at all?
>> 2. If not, is this a good solution to the problem?
>
> I think (not sure) another problem is the CSeq - which will be increased
> with every request, and if the proxy P! inreases it on behalf of UAC1,
> then this might cause problems.

IE you are saying one has to increment (well, anyway, change) CSeq so that
INVITE can be distinguished by P2/UAS from a delayed retransmit, yes? Which 
is what is meant by:

      If a proxy were to resubmit a request adding a Proxy-Authorization
      header field value, it would need to increment the CSeq in the new
      request.  However, this would cause the UAC that submitted the
      original request to discard a response from the UAS, as the CSeq
      value would be different.

Which means as UAC will be expecting the remainder of the transaction to be
carrying the first CSeq number, you would have to rewrite CSeq back to the
original CSeq number for the responses, and rewrite the ACK etc. too the
other way. Which is starting to get rather dirty. Hmmm... I shall think
more on that.

Perhaps one option would be for P1, on reception of the 407, to
407 the INVITE *LOCALLY* (i.e. from P1) saying it had a stale nonce,
and effectively ignore the request for authorization from P2/UAS.
UAC would then retry the INVITE (new CSeq) with a new set of
credentials for P1, and P1, now having the challenge from P2/UAS in
relation to that CallID, it could insert the the Proxy-Authorization
header itself.

This does mean that UAC is still going to have to authenticate
the INVITE twice. But at least the other problems I mentioned
are solved.

>> 3. If not, why not, and what should I do instead?
>
> SP2 can authenticate your IP address (IP address of P1) and all requests
> from your IP address will be billed on your account. To avaid spoffing,
> UDP should be disabled on P2 and you should send your requests using TCP.

Sadly that does not much help (I don't think).

If P2 is (say) a PSTN voice gatweay, it doesn't take a lot of imagination
to work out how to do sufficient sequence number guessing to open
a channel and issue one invite. If the reward is n calls of infinite
length to an international premium-rate number, then it's going to
be worth trying. Remember once people said "this won't be a problem"
for BGP.

The only (sensible) solutions I can see other than the above tweaks
or relying on TCP and IP address (which I am loath to do as I've
seen it go wrong before) are:

a) IPSec / TLS - proper transport layer authentication - which is
   unfortunate as it is relatively CPU intensive; or

b) Proxy (on every transaction) simply inserts a header which is
   identifies the proxy, and then has a hash of some (shared) secret,
   the proxy name, and To:/From:/CSeq:, which the UA/other proxy then
   checks is there. But that requires a separate authentication scheme
   which I was trying to avoid.

Alex

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        Hisham Khartabil <hisham.khartabil@nokia.com>,
        Allison Mankin <mankin@psg.com>, Robert Sparks <rjsparks@nostrum.com>,
        John Peterson <jon.peterson@neustar.biz>,
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From: Rohan Mahy <rohan@cisco.com>
Date: Tue, 6 Apr 2004 10:53:32 -0700
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Subject: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26
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Hello All,

Please mark your calendars.

I'd like to announce a joint interim of the SIP, SIPPING, SIMPLE, and  
XCON WGs.
Nokia has generously agreed to host the event either at their offices  
near Boston or in a neighboring hotel during the week of May 24.

Please reserve the following dates for the WGs that you are interested  
in:
Monday May 24 SIP and SIPPING  tentatively 9 am start
Tuesday May 25 SIMPLE   All day
Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch  
flights.
(the actual agenda is subject to change, including the dates of various  
WG meetings).

There will be no fee, however please let Hisham and myself know if you  
are planning to attend by clicking on the Yes or Maybe links below:

mailto:rohan@cisco.com? 
Cc=hisham.khartabil@nokia.com&Subject=[interim]yes
mailto:rohan@cisco.com? 
Cc=hisham.khartabil@nokia.com&Subject=[interim]maybe

Wireless will be provided. As usual, I will be making T-shirts.  If you  
are interested in buying one, please send your shirt size to me.

More details on the agenda and hotel accommodations will be forthcoming  
shortly.

thanks,
-rohan


_______________________________________________
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: Tue, 06 Apr 2004 14:12:20 -0600
To: Rohan Mahy <rohan@cisco.com>, "'sip@ietf.org'" <sip@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26
Cc: Dean Willis <dean.willis@softarmor.com>,
        Hisham Khartabil <hisham.khartabil@nokia.com>,
        Allison Mankin <mankin@psg.com>, Robert Sparks <rjsparks@nostrum.com>,
        John Peterson <jon.peterson@neustar.biz>,
        Ted Hardie <hardie@qualcomm.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        Rohan Mahy <rohan@cisco.com>, Alan Johnston <alan.johnston@wcom.com>,
        Adam Roach <adam@dynamicsoft.com>
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There were enough open issues surrounding the emergency team/effort that 
I'm concerned that there might not be enough time to have all of SIP & 
SIPPING in one day (with respect to how much time we took just during the 
e2m discussions in Seoul (2.5 hours) - virtually all of which was useful).

I otherwise like the ordering of WGs in the tentative scheduling

At 10:53 AM 4/6/2004 -0700, Rohan Mahy wrote:
>Hello All,
>
>Please mark your calendars.
>
>I'd like to announce a joint interim of the SIP, SIPPING, SIMPLE, and
>XCON WGs.
>Nokia has generously agreed to host the event either at their offices
>near Boston or in a neighboring hotel during the week of May 24.
>
>Please reserve the following dates for the WGs that you are interested
>in:
>Monday May 24 SIP and SIPPING  tentatively 9 am start
>Tuesday May 25 SIMPLE   All day
>Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch
>flights.
>(the actual agenda is subject to change, including the dates of various
>WG meetings).
>
>There will be no fee, however please let Hisham and myself know if you
>are planning to attend by clicking on the Yes or Maybe links below:
>
>mailto:rohan@cisco.com? Cc=hisham.khartabil@nokia.com&Subject=[interim]yes
>mailto:rohan@cisco.com? Cc=hisham.khartabil@nokia.com&Subject=[interim]maybe
>
>Wireless will be provided. As usual, I will be making T-shirts.  If you
>are interested in buying one, please send your shirt size to me.
>
>More details on the agenda and hotel accommodations will be forthcoming
>shortly.
>
>thanks,
>-rohan
>
>
>_______________________________________________
>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


_______________________________________________
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: "'James M. Polk'" <jmpolk@cisco.com>, Rohan Mahy <rohan@cisco.com>,
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Cc: Dean Willis <dean.willis@softarmor.com>,
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        Rohan Mahy
	 <rohan@cisco.com>, Alan Johnston <alan.johnston@wcom.com>,
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Subject: RE: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26
Date: Tue, 6 Apr 2004 16:10:48 -0500 
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I also hate to say that it does look like a tight schedule given current
issues and past history of interim meetings.  Previous Interims for
SIP/SIPPING were 2 full days and SIMPLE last year was also 2 full days. I
don't think anyone would suggest the time was not well utilized at any of
those meetings.  Since the start day has been moved to Monday (rather than
the initial proposal of Tuesday), is there a possibility of continuing into
Thursday? 

Mary.

-----Original Message-----
From: James M. Polk [mailto:jmpolk@cisco.com] 
Sent: Tuesday, April 06, 2004 3:12 PM
To: Rohan Mahy; 'sip@ietf.org'
Cc: Dean Willis; Hisham Khartabil; Allison Mankin; Robert Sparks; John
Peterson; Ted Hardie; Gonzalo Camarillo; Rohan Mahy; Alan Johnston; Adam
Roach
Subject: Re: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26


There were enough open issues surrounding the emergency team/effort that 
I'm concerned that there might not be enough time to have all of SIP & 
SIPPING in one day (with respect to how much time we took just during the 
e2m discussions in Seoul (2.5 hours) - virtually all of which was useful).

I otherwise like the ordering of WGs in the tentative scheduling

At 10:53 AM 4/6/2004 -0700, Rohan Mahy wrote:
>Hello All,
>
>Please mark your calendars.
>
>I'd like to announce a joint interim of the SIP, SIPPING, SIMPLE, and 
>XCON WGs. Nokia has generously agreed to host the event either at their 
>offices near Boston or in a neighboring hotel during the week of May 
>24.
>
>Please reserve the following dates for the WGs that you are interested
>in:
>Monday May 24 SIP and SIPPING  tentatively 9 am start
>Tuesday May 25 SIMPLE   All day
>Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch 
>flights. (the actual agenda is subject to change, including the dates 
>of various WG meetings).
>
>There will be no fee, however please let Hisham and myself know if you 
>are planning to attend by clicking on the Yes or Maybe links below:
>
>mailto:rohan@cisco.com? 
>Cc=hisham.khartabil@nokia.com&Subject=[interim]yes
>mailto:rohan@cisco.com?
Cc=hisham.khartabil@nokia.com&Subject=[interim]maybe
>
>Wireless will be provided. As usual, I will be making T-shirts.  If you 
>are interested in buying one, please send your shirt size to me.
>
>More details on the agenda and hotel accommodations will be forthcoming 
>shortly.
>
>thanks,
>-rohan
>
>
>_______________________________________________
>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


_______________________________________________
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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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26</TITLE>
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<BODY>

<P><FONT SIZE=3D2>I also hate to say that it does look like a tight =
schedule given current issues and past history of interim =
meetings.&nbsp; Previous Interims for SIP/SIPPING were 2 full days and =
SIMPLE last year was also 2 full days. I don't think anyone would =
suggest the time was not well utilized at any of those meetings.&nbsp; =
Since the start day has been moved to Monday (rather than the initial =
proposal of Tuesday), is there a possibility of continuing into =
Thursday? </FONT></P>

<P><FONT SIZE=3D2>Mary.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: James M. Polk [<A =
HREF=3D"mailto:jmpolk@cisco.com">mailto:jmpolk@cisco.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, April 06, 2004 3:12 PM</FONT>
<BR><FONT SIZE=3D2>To: Rohan Mahy; 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>Cc: Dean Willis; Hisham Khartabil; Allison Mankin; =
Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo; Rohan =
Mahy; Alan Johnston; Adam Roach</FONT></P>

<P><FONT SIZE=3D2>Subject: Re: [Sip] Joint Interim Meeting Mon May 24 - =
Wed May 26</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>There were enough open issues surrounding the =
emergency team/effort that </FONT>
<BR><FONT SIZE=3D2>I'm concerned that there might not be enough time to =
have all of SIP &amp; </FONT>
<BR><FONT SIZE=3D2>SIPPING in one day (with respect to how much time we =
took just during the </FONT>
<BR><FONT SIZE=3D2>e2m discussions in Seoul (2.5 hours) - virtually all =
of which was useful).</FONT>
</P>

<P><FONT SIZE=3D2>I otherwise like the ordering of WGs in the tentative =
scheduling</FONT>
</P>

<P><FONT SIZE=3D2>At 10:53 AM 4/6/2004 -0700, Rohan Mahy wrote:</FONT>
<BR><FONT SIZE=3D2>&gt;Hello All,</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Please mark your calendars.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;I'd like to announce a joint interim of the SIP, =
SIPPING, SIMPLE, and </FONT>
<BR><FONT SIZE=3D2>&gt;XCON WGs. Nokia has generously agreed to host =
the event either at their </FONT>
<BR><FONT SIZE=3D2>&gt;offices near Boston or in a neighboring hotel =
during the week of May </FONT>
<BR><FONT SIZE=3D2>&gt;24.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Please reserve the following dates for the WGs =
that you are interested</FONT>
<BR><FONT SIZE=3D2>&gt;in:</FONT>
<BR><FONT SIZE=3D2>&gt;Monday May 24 SIP and SIPPING&nbsp; tentatively =
9 am start</FONT>
<BR><FONT SIZE=3D2>&gt;Tuesday May 25 SIMPLE&nbsp;&nbsp; All day</FONT>
<BR><FONT SIZE=3D2>&gt;Wednesday May 26 XCON&nbsp; tentatively finish =
at 4pm so folks can catch </FONT>
<BR><FONT SIZE=3D2>&gt;flights. (the actual agenda is subject to =
change, including the dates </FONT>
<BR><FONT SIZE=3D2>&gt;of various WG meetings).</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;There will be no fee, however please let Hisham =
and myself know if you </FONT>
<BR><FONT SIZE=3D2>&gt;are planning to attend by clicking on the Yes or =
Maybe links below:</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;<A =
HREF=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>? </FONT>
<BR><FONT =
SIZE=3D2>&gt;Cc=3Dhisham.khartabil@nokia.com&amp;Subject=3D[interim]yes<=
/FONT>
<BR><FONT SIZE=3D2>&gt;<A =
HREF=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>? =
Cc=3Dhisham.khartabil@nokia.com&amp;Subject=3D[interim]maybe</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;Wireless will be provided. As usual, I will be =
making T-shirts.&nbsp; If you </FONT>
<BR><FONT SIZE=3D2>&gt;are interested in buying one, please send your =
shirt size to me.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;More details on the agenda and hotel =
accommodations will be forthcoming </FONT>
<BR><FONT SIZE=3D2>&gt;shortly.</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;thanks,</FONT>
<BR><FONT SIZE=3D2>&gt;-rohan</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT =
SIZE=3D2>&gt;_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt;Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt;This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>&gt;Use sip-implementors@cs.columbia.edu for =
questions on current sip Use </FONT>
<BR><FONT SIZE=3D2>&gt;sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>cheers,</FONT>
<BR><FONT SIZE=3D2>James</FONT>
</P>

<P><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
*******************</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Truth is not to be argued... it is to =
be presented</FONT>
</P>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip Use sipping@ietf.org for new developments on the =
application of sip</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
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From exim@www1.ietf.org  Tue Apr  6 22:21:59 2004
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Message-ID: <31F31018FE2DC941A70BD30477510B0C19D644@biscm02>
From: "Hu Yijun,BISC R&D SA(BJ)" <yijun.hu@siemens.com>
To: "'Umesh Sharma'" <umesh_sharma@persistent.co.in>,
        "Hu Yijun,BISC R&D SA(BJ)" <yijun.hu@siemens.com>, sip@ietf.org
Subject: RE: [SIP] Sending  2xx responses to INVITE
Date: Wed, 7 Apr 2004 08:46:38 +0800 
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hello, Umesh

The more detailed statement to the sever transaction is described in ch17.2.1:

"   If, while in the "Proceeding" state, the TU passes a 2xx response to
   the server transaction, the server transaction MUST pass this
   response to the transport layer for transmission.  It is not
   retransmitted by the server transaction; retransmissions of 2xx
   responses are handled by the TU.  The server transaction MUST then
   transition to the "Terminated" state.

   While in the "Proceeding" state, if the TU passes a response with
   status code from 300 to 699 to the server transaction, the response
   MUST be passed to the transport layer for transmission, and the state
   machine MUST enter the "Completed" state.  For unreliable transports,
   timer G is set to fire in T1 seconds, and is not set to fire for
   reliable transports."

BR
hu yijun

>-----Original Message-----
>From: Umesh Sharma [mailto:umesh_sharma@persistent.co.in]
>Sent: Tuesday, April 06, 2004 8:40 PM
>To: Hu Yijun,BISC R&D SA(BJ); sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>hi Hu
>
>I re-read the section 17.1.1.1, it looks to me a general 
>statement which
>does not
>distinguishes between non-2xx final & 2xx response.
>
>-regards
>umesh
>
>-----Original Message-----
>From: Hu Yijun,BISC R&D SA(BJ) [mailto:yijun.hu@siemens.com]
>Sent: Tuesday, April 06, 2004 6:32 AM
>To: 'Umesh Sharma'; Bilajbegovic Damir; ??????????? ????? ??????????;
>sip@ietf.org
>Subject: RE: [SIP] Sending 2xx responses to INVITE
>
>
>Hello,
>
>Please see my comments below.
>
>regards
>hu yijun
>
>>-----Original Message-----
>>From: Umesh Sharma [mailto:umesh_sharma@persistent.co.in]
>>Sent: Monday, April 05, 2004 8:06 PM
>>To: Bilajbegovic Damir; ??????????? ????? ??????????; sip@ietf.org
>>Subject: RE: [SIP] Sending 2xx responses to INVITE
>>
>>
>>Hi
>>
>>>>In what way does TU send retransmissions of 2xx responses to
>>INVITE? What
>>is
>>>>the interval between these retransmissions?
>>
>>To quote from rfc 3261 directly:
>>
>>"The 2xx response is passed to the transport with an
>> interval that starts at T1 seconds and doubles for each
>> retransmission until it reaches T2 seconds (T1 and T2 are defined in
>> Section 17).  Response retransmissions cease when an ACK request for
>> the response is received.  This is independent of whatever transport
>> protocols are used to send the response."
>>
>>>>How many retansmissions can be sent?
>>
>>"If the server retransmits the 2xx response for 64*T1 seconds without
>> receiving an ACK, the dialog is confirmed, but the session SHOULD be
>> terminated."
>>
>>In 17.1.1.1
>>
>>"Eventually, the server transaction decides
>>   to send a final response.  For unreliable transports, that response
>>   is retransmitted periodically, and for reliable transports, it is
>>   sent once."
>>
>>I have a question too here, Now this line above seems to
>>contradict this
>>paragraph in
>>13.3.1.4.
>>
>>      "Since 2xx is retransmitted end-to-end, there may be 
>hops between
>>      UAS and UAC that are UDP.  To ensure reliable delivery across
>>      these hops, the response is retransmitted periodically
>>even if the
>>      transport at the UAS is reliable."
>>
>>Am I reading this incorrectly?
>
>The description of ch 17.1.1.1 was foucs on the retransmission of the
>non-2xx final response to INVITE, which were administered by 
>the transaction
>layer. Those responses are delivered hop-by-hop. So that, for 
>a certain hop,
>which uses reliable transport, the non-2xx final response is 
>sent only once.
>
>However, the retransmission of the 2XX to INVITE was 
>administered by the
>transaction user (TU). It MUST comply with the ch13.3.1.4 of 
>rfc3261: "the
>response is retransmitted periodically even if the transport 
>at the UAS is
>reliable".
>
>>
>>regards
>>-umesh
>>-----Original Message-----
>>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
>>Bilajbegovic Damir
>>Sent: Monday, April 05, 2004 4:18 PM
>>To: ??????????? ????? ??????????; sip@ietf.org
>>Subject: RE: [SIP] Sending 2xx responses to INVITE
>>
>>
>>Hi,
>>	Not sure but I think 1st Invite and 7 retransmission.
>>Intervals are
>>increasing by double.
>>		Hope I helped.
>>			Bilaj
>>
>>-----Original Message-----
>>From: galactionov@loniis.ru [mailto:galactionov@loniis.ru]
>>Sent: Monday, April 05, 2004 12:34 PM
>>To: sip@ietf.org
>>Subject: [SIP] Sending 2xx responses to INVITE
>>
>>
>>Dear SIP experts,
>>
>>I have following questions (maybe already discussed on the
>>list in the past)
>>regarding the sending of 2xx responses to INVITE.
>>
>>In what way does TU send retransmissions of 2xx responses to
>>INVITE? What is
>>the interval between these retransmissions? How many
>>retansmissions can be
>>sent?
>>
>>Best regards
>>Anton
>>
>>
>>_______________________________________________
>>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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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] Joint Interim Meeting Mon May 24 - Wed May 26
Date: Tue, 6 Apr 2004 19:43:07 -0500
Message-ID: <28F05913385EAC43AF019413F674A01708145A5C@OCCLUST04EVS1.ugd.att.com>
Thread-Topic: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26
Thread-Index: AcQcK9xl10z4GozqQ1SEWefKTdWUZQADMZrg
From: "Dolly, Martin C, ALABS" <mdolly@att.com>
To: "Mary Barnes" <mary.barnes@nortelnetworks.com>,
        "James M. Polk" <jmpolk@cisco.com>, "Rohan Mahy" <rohan@cisco.com>,
        <sip@ietf.org>
Cc: "Dean Willis" <dean.willis@softarmor.com>,
        "Hisham Khartabil" <hisham.khartabil@nokia.com>,
        "Allison Mankin" <mankin@psg.com>,
        "Robert Sparks" <rjsparks@nostrum.com>,
        "John Peterson" <jon.peterson@neustar.biz>,
        "Ted Hardie" <hardie@qualcomm.com>,
        "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>,
        "Rohan Mahy" <rohan@cisco.com>,
        "Alan Johnston" <alan.johnston@wcom.com>,
        "Adam Roach" <adam@dynamicsoft.com>
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I believe that 2 days for SIP/SIPPING would be more approriate
=20
Cheers,
=20
Martin

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Mary
Barnes
Sent: Tuesday, April 06, 2004 5:11 PM
To: 'James M. Polk'; Rohan Mahy; 'sip@ietf.org'
Cc: Dean Willis; Hisham Khartabil; Allison Mankin; Robert Sparks; John
Peterson; Ted Hardie; Gonzalo Camarillo; Rohan Mahy; Alan Johnston; Adam
Roach
Subject: RE: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26



I also hate to say that it does look like a tight schedule given current
issues and past history of interim meetings.  Previous Interims for
SIP/SIPPING were 2 full days and SIMPLE last year was also 2 full days.
I don't think anyone would suggest the time was not well utilized at any
of those meetings.  Since the start day has been moved to Monday (rather
than the initial proposal of Tuesday), is there a possibility of
continuing into Thursday?=20

Mary.=20

-----Original Message-----=20
From: James M. Polk [ mailto:jmpolk@cisco.com]=20
Sent: Tuesday, April 06, 2004 3:12 PM=20
To: Rohan Mahy; 'sip@ietf.org'=20
Cc: Dean Willis; Hisham Khartabil; Allison Mankin; Robert Sparks; John
Peterson; Ted Hardie; Gonzalo Camarillo; Rohan Mahy; Alan Johnston; Adam
Roach

Subject: Re: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26=20


There were enough open issues surrounding the emergency team/effort that

I'm concerned that there might not be enough time to have all of SIP &=20
SIPPING in one day (with respect to how much time we took just during
the=20
e2m discussions in Seoul (2.5 hours) - virtually all of which was
useful).=20

I otherwise like the ordering of WGs in the tentative scheduling=20

At 10:53 AM 4/6/2004 -0700, Rohan Mahy wrote:=20
>Hello All,=20
>=20
>Please mark your calendars.=20
>=20
>I'd like to announce a joint interim of the SIP, SIPPING, SIMPLE, and=20
>XCON WGs. Nokia has generously agreed to host the event either at their

>offices near Boston or in a neighboring hotel during the week of May=20
>24.=20
>=20
>Please reserve the following dates for the WGs that you are interested=20
>in:=20
>Monday May 24 SIP and SIPPING  tentatively 9 am start=20
>Tuesday May 25 SIMPLE   All day=20
>Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch=20
>flights. (the actual agenda is subject to change, including the dates=20
>of various WG meetings).=20
>=20
>There will be no fee, however please let Hisham and myself know if you=20
>are planning to attend by clicking on the Yes or Maybe links below:=20
>=20
> mailto:rohan@cisco.com?=20
>Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes=20
> mailto:rohan@cisco.com?
Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe=20
>=20
>Wireless will be provided. As usual, I will be making T-shirts.  If you

>are interested in buying one, please send your shirt size to me.=20
>=20
>More details on the agenda and hotel accommodations will be forthcoming

>shortly.=20
>=20
>thanks,=20
>-rohan=20
>=20
>=20
>_______________________________________________=20
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip=20
>This list is for NEW development of the core SIP Protocol=20
>Use sip-implementors@cs.columbia.edu for questions on current sip Use=20
>sipping@ietf.org for new developments on the application of sip=20


cheers,=20
James=20

                                *******************=20
                 Truth is not to be argued... it is to be presented=20


_______________________________________________=20
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip=20
This list is for NEW development of the core SIP Protocol=20
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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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>RE: [Sip] Joint Interim Meeting Mon May 24 - Wed May 26</TITLE>

<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D905243800-07042004><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
believe that 2 days for SIP/SIPPING would be more =
approriate</FONT></SPAN></DIV>
<DIV><SPAN class=3D905243800-07042004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D905243800-07042004><FONT face=3DArial color=3D#0000ff =

size=3D2>Cheers,</FONT></SPAN></DIV>
<DIV><SPAN class=3D905243800-07042004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D905243800-07042004><FONT face=3DArial color=3D#0000ff =

size=3D2>Martin</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Mary =
Barnes<BR><B>Sent:</B>=20
  Tuesday, April 06, 2004 5:11 PM<BR><B>To:</B> 'James M. Polk'; Rohan =
Mahy;=20
  'sip@ietf.org'<BR><B>Cc:</B> Dean Willis; Hisham Khartabil; Allison =
Mankin;=20
  Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo; Rohan =
Mahy; Alan=20
  Johnston; Adam Roach<BR><B>Subject:</B> RE: [Sip] Joint Interim =
Meeting Mon=20
  May 24 - Wed May 26<BR><BR></FONT></DIV>
  <P><FONT size=3D2>I also hate to say that it does look like a tight =
schedule=20
  given current issues and past history of interim meetings.&nbsp; =
Previous=20
  Interims for SIP/SIPPING were 2 full days and SIMPLE last year was =
also 2 full=20
  days. I don't think anyone would suggest the time was not well =
utilized at any=20
  of those meetings.&nbsp; Since the start day has been moved to Monday =
(rather=20
  than the initial proposal of Tuesday), is there a possibility of =
continuing=20
  into Thursday? </FONT></P>
  <P><FONT size=3D2>Mary.</FONT> </P>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: James=20
  M. Polk [<A =
href=3D"mailto:jmpolk@cisco.com">mailto:jmpolk@cisco.com</A>]=20
  </FONT><BR><FONT size=3D2>Sent: Tuesday, April 06, 2004 3:12 PM</FONT> =
<BR><FONT=20
  size=3D2>To: Rohan Mahy; 'sip@ietf.org'</FONT> <BR><FONT size=3D2>Cc: =
Dean Willis;=20
  Hisham Khartabil; Allison Mankin; Robert Sparks; John Peterson; Ted =
Hardie;=20
  Gonzalo Camarillo; Rohan Mahy; Alan Johnston; Adam Roach</FONT></P>
  <P><FONT size=3D2>Subject: Re: [Sip] Joint Interim Meeting Mon May 24 =
- Wed May=20
  26</FONT> </P><BR>
  <P><FONT size=3D2>There were enough open issues surrounding the =
emergency=20
  team/effort that </FONT><BR><FONT size=3D2>I'm concerned that there =
might not be=20
  enough time to have all of SIP &amp; </FONT><BR><FONT size=3D2>SIPPING =
in one=20
  day (with respect to how much time we took just during the =
</FONT><BR><FONT=20
  size=3D2>e2m discussions in Seoul (2.5 hours) - virtually all of which =
was=20
  useful).</FONT> </P>
  <P><FONT size=3D2>I otherwise like the ordering of WGs in the =
tentative=20
  scheduling</FONT> </P>
  <P><FONT size=3D2>At 10:53 AM 4/6/2004 -0700, Rohan Mahy wrote:</FONT> =
<BR><FONT=20
  size=3D2>&gt;Hello All,</FONT> <BR><FONT size=3D2>&gt;</FONT> =
<BR><FONT=20
  size=3D2>&gt;Please mark your calendars.</FONT> <BR><FONT =
size=3D2>&gt;</FONT>=20
  <BR><FONT size=3D2>&gt;I'd like to announce a joint interim of the =
SIP, SIPPING,=20
  SIMPLE, and </FONT><BR><FONT size=3D2>&gt;XCON WGs. Nokia has =
generously agreed=20
  to host the event either at their </FONT><BR><FONT =
size=3D2>&gt;offices near=20
  Boston or in a neighboring hotel during the week of May =
</FONT><BR><FONT=20
  size=3D2>&gt;24.</FONT> <BR><FONT size=3D2>&gt;</FONT> <BR><FONT =
size=3D2>&gt;Please=20
  reserve the following dates for the WGs that you are interested</FONT> =

  <BR><FONT size=3D2>&gt;in:</FONT> <BR><FONT size=3D2>&gt;Monday May 24 =
SIP and=20
  SIPPING&nbsp; tentatively 9 am start</FONT> <BR><FONT =
size=3D2>&gt;Tuesday May=20
  25 SIMPLE&nbsp;&nbsp; All day</FONT> <BR><FONT size=3D2>&gt;Wednesday =
May 26=20
  XCON&nbsp; tentatively finish at 4pm so folks can catch =
</FONT><BR><FONT=20
  size=3D2>&gt;flights. (the actual agenda is subject to change, =
including the=20
  dates </FONT><BR><FONT size=3D2>&gt;of various WG meetings).</FONT> =
<BR><FONT=20
  size=3D2>&gt;</FONT> <BR><FONT size=3D2>&gt;There will be no fee, =
however please=20
  let Hisham and myself know if you </FONT><BR><FONT size=3D2>&gt;are =
planning to=20
  attend by clicking on the Yes or Maybe links below:</FONT> <BR><FONT=20
  size=3D2>&gt;</FONT> <BR><FONT size=3D2>&gt;<A=20
  href=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>? =
</FONT><BR><FONT=20
  =
size=3D2>&gt;Cc=3Dhisham.khartabil@nokia.com&amp;Subject=3D[interim]yes</=
FONT>=20
  <BR><FONT size=3D2>&gt;<A=20
  href=3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com</A>?=20
  Cc=3Dhisham.khartabil@nokia.com&amp;Subject=3D[interim]maybe</FONT> =
<BR><FONT=20
  size=3D2>&gt;</FONT> <BR><FONT size=3D2>&gt;Wireless will be provided. =
As usual, I=20
  will be making T-shirts.&nbsp; If you </FONT><BR><FONT =
size=3D2>&gt;are=20
  interested in buying one, please send your shirt size to me.</FONT> =
<BR><FONT=20
  size=3D2>&gt;</FONT> <BR><FONT size=3D2>&gt;More details on the agenda =
and hotel=20
  accommodations will be forthcoming </FONT><BR><FONT =
size=3D2>&gt;shortly.</FONT>=20
  <BR><FONT size=3D2>&gt;</FONT> <BR><FONT size=3D2>&gt;thanks,</FONT> =
<BR><FONT=20
  size=3D2>&gt;-rohan</FONT> <BR><FONT size=3D2>&gt;</FONT> <BR><FONT=20
  size=3D2>&gt;</FONT> <BR><FONT=20
  size=3D2>&gt;_______________________________________________</FONT> =
<BR><FONT=20
  size=3D2>&gt;Sip mailing list&nbsp; <A=20
  href=3D"https://www1.ietf.org/mailman/listinfo/sip"=20
  target=3D_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> =
<BR><FONT=20
  size=3D2>&gt;This list is for NEW development of the core SIP =
Protocol</FONT>=20
  <BR><FONT size=3D2>&gt;Use sip-implementors@cs.columbia.edu for =
questions on=20
  current sip Use </FONT><BR><FONT size=3D2>&gt;sipping@ietf.org for new =

  developments on the application of sip</FONT> </P><BR>
  <P><FONT size=3D2>cheers,</FONT> <BR><FONT size=3D2>James</FONT> </P>
  <P><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  *******************</FONT> <BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Truth is not to be argued... it is to be presented</FONT> </P><BR>
  <P><FONT =
size=3D2>_______________________________________________</FONT>=20
  <BR><FONT size=3D2>Sip mailing list&nbsp; <A=20
  href=3D"https://www1.ietf.org/mailman/listinfo/sip"=20
  target=3D_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> =
<BR><FONT=20
  size=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>=20
  <BR><FONT size=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current=20
  sip Use sipping@ietf.org for new developments on the application of=20
  sip</FONT></P></BLOCKQUOTE></BODY></HTML>

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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



From exim@www1.ietf.org  Wed Apr  7 02:37:00 2004
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hello,all.
i will kick off my master study.
the research area is about sip in 3G mobile network.
anyone can give some advices about it. Give me some direction.
thanks.

_________________________________________________________________
=D3=EB=C1=AA=BB=FA=B5=C4=C5=F3=D3=D1=BD=F8=D0=D0=BD=BB=C1=F7=A3=AC=C7=EB=CA=
=B9=D3=C3 MSN Messenger:  http://messenger.msn.com/cn =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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Date: Mon, 05 Apr 2004 17:11:21 -0400
Subject: [Sip] Last Call: 'Session Initiation Protocol (SIP) Extension for Event
 State Publication' to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol WG to 
consider the following document:

- 'Session Initiation Protocol (SIP) Extension for Event State Publication '
   <draft-ietf-sip-publish-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 2004-04-19.

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


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Subject: [Sip] Last Call: 'S/MIME AES Requirement for SIP' to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol WG to 
consider the following document:

- 'S/MIME AES Requirement for SIP '
   <draft-ietf-sip-smime-aes-01.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
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iesg@ietf.org or ietf@ietf.org mailing lists by 2004-04-19.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-smime-aes-01.txt


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From exim@www1.ietf.org  Wed Apr  7 05:50:04 2004
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Subject: [Sip] Last Call: 'The Session Inititation Protocol (SIP) 'Join'
 Header' to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol 
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- 'The Session Inititation Protocol (SIP) 'Join' Header '
   <draft-ietf-sip-join-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 2004-04-19.

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


_______________________________________________
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From exim@www1.ietf.org  Wed Apr  7 14:04:17 2004
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Date: Wed, 07 Apr 2004 16:41:44 +0100
From: Alex Bligh <alex@alex.org.uk>
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To: Fredrik Thulin <ft@it.su.se>,
        Klaus Darilion <klaus.mailinglists@pernau.at>
Cc: sip@ietf.org, Alex Bligh <alex@alex.org.uk>
Subject: Re: [Sip] RFC3261 & authenticating PROXIES to other proxies & UAS's
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--On 06 April 2004 18:39 +0200 Fredrik Thulin <ft@it.su.se> wrote:

> There is wording in section 22.3 saying proxys MUST NOT add things to
> authorization headers, and that all 407 responses MUST be forwarded
> upstreams. That is a pity, since I think otherwise the following could
> solve  the problem :
>
> P1 could, upon receiving the 407 from P2, start a new transaction with a
> different branch parameter and added authentication credentials. The
> request  could then be re-submitted to P2.

If I read things right, the branch parameter is going to be different
/anyway/ for a proxy with loop detection:

16.6 Request Forwarding [Section 8]

...
         If a proxy wishes to detect loops, the "branch" parameter it
         supplies MUST depend on all information affecting processing of
         a request, including the incoming Request-URI and any header
         fields affecting the request's admission or routing.  This is
         necessary to distinguish looped requests from requests whose
         routing parameters have changed before returning to this
         server.

On this basis, if the proxy retries the INVITE with the same CSeq,
what is the problem? IE how is this any different from spiraling
(which is allowed)?


Alex

_______________________________________________
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From exim@www1.ietf.org  Wed Apr  7 14:19:43 2004
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Date: Wed, 07 Apr 2004 17:11:52 +0100
From: Alex Bligh <alex@alex.org.uk>
Reply-To: Alex Bligh <alex@alex.org.uk>
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Subject: [Sip] Unknown via-params
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SHOULD / MUST SIP devices ignore/strip/reject via-params that are not
defined by RFC3261?

IE if I build a proxy which inserts a proprietary via-param (so
for instance inserts a Via line like):

Via: SIP/2.0/UDP host.user.com:4321;
     maddr=224.1.2.3;branch=f4fs34.1;myparam=foobar

can I guarantee that compliant proxies & UAs will ignore (rather
than reject, or, in the case of a proxy strip) the myparam=foobar?

Alex


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From exim@www1.ietf.org  Wed Apr  7 22:36:44 2004
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Reply-To: <brett@broadsoft.com>
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Subject: RE: [Sip] Unknown via-params
Date: Wed, 7 Apr 2004 19:13:45 -0400
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> SHOULD / MUST SIP devices ignore/strip/reject 
> via-params that are not defined by RFC3261?

The receiver MUST echo them back.  However 
extra Via parameters were not part of rfc2543.  
Thus some older devices might strip the extra 
parameters or reject the message.


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From exim@www1.ietf.org  Thu Apr  8 12:54:49 2004
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From: Viktor.Bekesi@nokia.com
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Subject: [Sip] Caller-Preferences: Representing negated Boolean feature tags
Date: Thu, 8 Apr 2004 13:28:24 +0200
Message-ID: <DF3C159C4F4BCE4BB35B43F4AB6D3D8401BC7ED0@buebe002.europe.nokia.com>
thread-topic: [Sip] Caller-Preferences: Representing negated Boolean feature tags
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Hello,

I am implementing SIP Caller-Preferences related functionalities, and =
there is one thing that is not clear for me after reading the relevant =
drafts and rfcs: Does it make sense to negate a Boolean type feature =
tag? like the audio in the example:

   (& (! (audio=3DTRUE))
      (video=3DTRUE)
      (sip.mobility=3Dfixed)
      (message=3DTRUE)
      (| (sip.methods=3DINVITE) (sip.methods=3DOPTIONS) =
(sip.methods=3DBYE)
         (sip.methods=3DCANCEL) (sip.methods=3DACK))
      (| (sip.schemes=3Dsip) (sip.schemes=3Dhttp)))


If yes, then the next question arises: How can I represent negated =
Boolean type feature tags in SIP header parameters?=20
According to draft-ietf-sip-callerprefs-10,=20

   (& (sip.mobility=3Dfixed)
      (| (! (sip.events=3Dpresence)) (sip.events=3Dwinfo))
      (| (language=3Den) (language=3Dde))
      (sip.description=3D"PC")
      (sip.newparam=3DTRUE)
      (rangeparam=3D-4..5125/1000))

this declaration can be turned into this representation:

   =
Accept-Contact:*;mobility=3D"fixed";events=3D"!presence,winfo";language=3D=
"en,de"
    ;description=3D"<PC>";+sip.newparam;+rangeparam=3D"#-4:+5.125"

(I reversed the order as it was in the document)

Here the events are negated and that is indicated by the exclamation =
mark at the beginning of the event-tag-value. Boolean type tags are =
turned into flag parameters with no value. So then how can the very =
first example represented?


Thanks in advance, best regards:=20

Viktor Bekesi
Nokia Networks
+36-20 9849-202
=20

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Date: Thu, 08 Apr 2004 13:49:17 +0100
From: Alex Bligh <alex@alex.org.uk>
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Subject: [Sip] RFC3261 and authenticating PROXIES to other proxies & UAS - 2
 possible solutions
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I am still trying to work out how one might authenticate that a call has
been transmitted by a proxy that the headers claimed transmitted it. SIP
provides facilities (in protocol) for proxies and UAS's to determine that a
call originated at a UAC with credentials, but no manner of determining
whether the call passed through a proxy with credentials. Whilst in some
circumstances this can be handled with network layer or transport layer
security, (a) this requires support on a hop by hop basis from the proxy
downstream, (b) the overhead may be substantial, and (c) it is somewhat
inconsistent that SIP supports this type of authentication for the UAC but
not proxies.

I've come up with 2 mechanisms to provide the support - one I posted
earlier (which requires only a change on the proxy with credentials) and
goes into the "Despite being against RFC3261, I can't see what this would
break" category, and the second (which requires a change both at the proxy
supplying credentials and at the server authenticating them) certainly
breaks nothing, but is less functional.

I'd appreciate comments, particularly on what the first mechanism breaks.
I (now) know that it is (today) against various bits of RFC3261. But I
cannot see what in practice it would break. And as a couple of people
(on and off list) have pointed out, it would be very useful (even for
reasons I hadn't thought of).

BACKGROUND
==========

The (general) problem I am trying to solve looks like this:


   UAC <-----/\/\/\/\/---> P1 <--/\/--> P2 <---> UAS => PSTN
  (CPE)                 (proxy)      (proxy)  (media
                                              gateway)

P2 may or may not be present.

Let us assume UAC and P1 are very distant, and P1 & either P2 or UAS are
sufficiently distant that host security is impractical without adding some
additional layer (ipsec, tls etc.).

UAC is CPE on a site of a customer of SP1, who operates P1. UAS is a media
gateway, and P2 a (possibly) intervening proxy, operated by SP2.

Let us further assume that UAC registers to P1, and that P1 has already
authenticated UAC, and that any INVITE passed through P1 has already been
authenticated as coming from UAC and been authorized by P1 according to
SP1's local policy, prior to being forwarded to P2.

Let us further assume that it is the local policy of SP1 that any such
authorized calls it passes to SP2 would be authorized by SP1. So for
instance SP2 is a VoIP termination provider, and SP1 a VoIP service
provider.

SP2 wishes to authenticate & authorize calls made to UAS. It uses some form
of authentication. What it in essence is trying to authenticate is that the
call actually came from P1, not that the call actually came from UAC. This
could either be done at P2, or at UAS.

I have two mechanisms of doing this

MECHANISM (AB)USING HTTP DIGEST AUTHENTICATION
==============================================

As per previous email, (ab)use HTTP digest authentication so
that the proxy retries the Invite with the same CSeq but with
a different branch parameter. This seems to me indistinguishable
from a spiraling scenario where the proxy concerned happens to
call the same Request-URI twice but other information in the message
header (in this case authentication) that may be used to route the
call has changed.

As such, I can't see what it would break, and it has the major
advantage it works with all existing (challenging) UA's & proxies,
and that being challenge-response, it is more secure than
other mechanisms.

But, for the sake of argument

ALTERNATIVE MECHANISM USING vauth Via-Parameter
===============================================

This mechanism introduces a new Via Parameter (vauth).

A UAC or Proxy inserting a Via Header with a RFC3261 compliant branch
parameter MAY introduce one or vauth parameter into the Via header, in each
case to identify authenticate the via header to one or more downstream
proxies or UAS's. Such UAS's or proxies MAY use a vauth header to take
local-policy dependent routing or access decisions. A UAC or Proxy which
does not insert a Via Header with a RFC3261 compliant branch parameter MUST
NOT insert any vauth parameters.

The vauth parameter is intended to allow a server to authenticate a subset
of the Via: header (being the sent-by entry and the branch token), together
with an opaque string, to downstream servers. Multiple vauth parameters MAY
be used in order to support multiple realms of authentication. The vauth
parameter supports extensible mechanisms of authentication, but the
mechanism specified herein is MD5 authentication using a shared secret.
The mechanism of exchange of the shared secret and authentication realm
is outside the scope of this document.

The vauthid field specifies the realm of authentication. The vauthid field
MUST be unique across instances if the vauth parameter in a single Via:
header. Those specifying vauth fields SHOULD ensure that they are globally
unique, for example a domain name MAY be used.

The vauthopaque field is an opaque field intended to pass information to
servers in the authentication realm which MAY be used to influence their
routing decisions - for instance it might be used to specify that access
should be permitted or denied, or have an accounting function.

The vauthmeth field specifies the mechanism of authentication. The only
currently defined method of authentication is MD5. In this mechanism, the
vauthmd5data field contains a hash of the sent-by field, the branch
parameter (which is, by RFC3261 unique across time and space), the vauthid
field, the vauthopaque field, the vauthexpire field (if any), and the
shared secret, which should be a string of printable characters no more
than 64 octets in length. The hash is calculated using MD5 (RFC 1321 [35]),
expressed in hexadecimal.

The optional vauthexpire field is a date in the SIP date format (i.e. the
preferred RFC1123 format) after which it should be assumed that the
header is not authentic. If a vauthexpire field is inserted, the server
MUST be synchronized to an accurate time source, and the expiry time
MUST be no shorter than 32 seconds beyond the time at which the Via:
header is to be transmitted.

A server MAY check the authenticity of a Via: header using vauth parameters
by performing each of the following checks:

1. Isolate the vauth field(s) corresponding to a realm for which the
   checking server possesses a shared secret.

2. If vauthexpire is present: if the checking server does not support
   vauthexpire, it MUST silently discard the request. If the checking
   server does support vauthexpire, it MUST check that the current
   time is not beyond the expiry time, and if it is, it MUST
   silently discard the request, else proceed.
   The silent discards are used to ensure that excessive
   transmission delays result only in the symptoms of packet loss,
   rather than false authentication refusals.

3. The server MUST recalculate the hash value and check whether
   it corresponds to the hash value in the vauthmd5data field.

4. The result of (3), the vauthopaque field, and the presence or
   absence of a vauthexpire field, MAY then be used
   to influence routing decisions

The vauth mechnanism is primarily designed to protect against spoofing
attacks, as the shared secret will not be known. It provides only limited
protection against replay attacks, in that replays are time-limited by the
vauth-expire header. Further limitation of replay attacks can be given by
the implementor using the vauthopaque field - for instance in an
environment where the Request-URI is known not to change between the server
inserting the Via header and the server checking it, the vauthopaque field
could be set to contain a hash of the Request-URI, which could be checked
upon receipt. Further, it should be noted that mechanism provides no
protection against malicious or accidental modification of the message
between transmission by the server inserting the Via: header and the server
analyzing it. It is not intended as a substitute for network or transport
layer security.

An example of a Via: header with vauth parameters follows:
Via: SIP/2/0/UDP 4.3.2.1:5060 branch=1fe452.12;
 vauth="server.example.com:5061,y,MD5=1234567890abcdef1234567890abcdef";
 vauth="server2.example.com:5060,,MD5=fedcba0987654321fedcba0987654321:Fri, 
13 Oct 1998 23:50:00 GMT"

RFC3261 specifies:

    via-params        =  via-ttl / via-maddr
                         / via-received / via-branch
                         / via-extension
    via-extension     =  generic-param
    generic-param     =  token [ EQUAL gen-value ]
    gen-value         =  token / host / quoted-string
    token             =  1*(alphanum / "-" / "." / "!" / "%" / "*"
                        / "_" / "+" / "`" / "'" / "~" )

This document introduces a new via-param which matches the generic-param
model:

    via-params        =  via-ttl / via-maddr
                         / via-received / via-branch
                         / via-vauth ; new
                         / via-extension
    via-vauth         =  "vauth" EQUAL vauth
    vauth             = LTQUOT vauthid COMMA vauthopaque
                        COMMA vauthmeth [ COLON vauthexpire ]RDQUOT
    vauthid           = vauthstr
    vauthopaque       = [ vauthstr ]
    vauthexpire       = SIP-date
    vauthstr          = 1*(alphanum / "-" / "." / "/" / ":" / "_" /
                        "[" / "]")
    vauthmeth         = vauthmeth-md5 / vauthmeth-ext
    vauthmeth-ext     = vauthextension EQUAL vauthextdata
    vauthextension    = 1*(alphanum)
    vauthextdat       = 1*(alphanum)
    vauthmeth-md5     = vauthmmd5 EQUAL vauthmd5data
    vauthmmd5         = "MD5"
    vauthmd5data      = 32LHEX


Alex

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To: Viktor.Bekesi@nokia.com
CC: sip@ietf.org
Subject: Re: [Sip] Caller-Preferences: Representing negated Boolean feature
 tags
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Viktor,

Sure, it makes sense to negate boolean feature tags. More below.

	Paul

Viktor.Bekesi@nokia.com wrote:
> Hello,
> 
> I am implementing SIP Caller-Preferences related functionalities, and there is one thing that is not clear for me after reading the relevant drafts and rfcs: Does it make sense to negate a Boolean type feature tag? like the audio in the example:
> 
>    (& (! (audio=TRUE))
>       (video=TRUE)
>       (sip.mobility=fixed)
>       (message=TRUE)
>       (| (sip.methods=INVITE) (sip.methods=OPTIONS) (sip.methods=BYE)
>          (sip.methods=CANCEL) (sip.methods=ACK))
>       (| (sip.schemes=sip) (sip.schemes=http)))

The first thing to decide is whether you want (! (audio=TRUE)), or 
(audio=FALSE). For some purposes they are the same, for others they are 
not. The difference comes when matching against another feature 
predicate that doesn't mention audio.

> If yes, then the next question arises: How can I represent negated Boolean type feature tags in SIP header parameters? 
> According to draft-ietf-sip-callerprefs-10, 
> 
>    (& (sip.mobility=fixed)
>       (| (! (sip.events=presence)) (sip.events=winfo))
>       (| (language=en) (language=de))
>       (sip.description="PC")
>       (sip.newparam=TRUE)
>       (rangeparam=-4..5125/1000))
> 
> this declaration can be turned into this representation:
> 
>    Accept-Contact:*;mobility="fixed";events="!presence,winfo";language="en,de"
>     ;description="<PC>";+sip.newparam;+rangeparam="#-4:+5.125"
 >
> (I reversed the order as it was in the document)
> 
> Here the events are negated and that is indicated by the exclamation mark at the beginning of the event-tag-value. Boolean type tags are turned into flag parameters with no value. So then how can the very first example represented?

The suppression of the TRUE value for boolean feature tags is syntactic 
sugar, and is optional. So the following are equivalent:

	audio
	audio="TRUE"

So, you can represent (! (audio=TRUE)) as

	audio="!TRUE"


> Thanks in advance, best regards: 
> 
> Viktor Bekesi
> Nokia Networks
> +36-20 9849-202
>  
> 
> _______________________________________________
> 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] Caller-Preferences: Representing negated Boolean feature tags
Date: Fri, 9 Apr 2004 09:07:41 +0200
Message-ID: <DF3C159C4F4BCE4BB35B43F4AB6D3D8401BC7ED9@buebe002.europe.nokia.com>
thread-topic: [Sip] Caller-Preferences: Representing negated Boolean feature tags
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Cc: <pkyzivat@cisco.com>
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Thank You for the answer.
Just one last question: (audio =3D FALSE) is equal to not including the =
audio tag at all?

Many thanks, BR: Viktor

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
Paul Kyzivat
Sent: Thursday April 08,2004 22:06
To: Bekesi Viktor (Nokia-NET/Budapest)
Cc: sip@ietf.org
Subject: Re: [Sip] Caller-Preferences: Representing negated Boolean
feature tags


Viktor,

Sure, it makes sense to negate boolean feature tags. More below.

	Paul

Viktor.Bekesi@nokia.com wrote:
> Hello,
>=20
> I am implementing SIP Caller-Preferences related functionalities, and =
there is one thing that is not clear for me after reading the relevant =
drafts and rfcs: Does it make sense to negate a Boolean type feature =
tag? like the audio in the example:
>=20
>    (& (! (audio=3DTRUE))
>       (video=3DTRUE)
>       (sip.mobility=3Dfixed)
>       (message=3DTRUE)
>       (| (sip.methods=3DINVITE) (sip.methods=3DOPTIONS) =
(sip.methods=3DBYE)
>          (sip.methods=3DCANCEL) (sip.methods=3DACK))
>       (| (sip.schemes=3Dsip) (sip.schemes=3Dhttp)))

The first thing to decide is whether you want (! (audio=3DTRUE)), or=20
(audio=3DFALSE). For some purposes they are the same, for others they =
are=20
not. The difference comes when matching against another feature=20
predicate that doesn't mention audio.

> If yes, then the next question arises: How can I represent negated =
Boolean type feature tags in SIP header parameters?=20
> According to draft-ietf-sip-callerprefs-10,=20
>=20
>    (& (sip.mobility=3Dfixed)
>       (| (! (sip.events=3Dpresence)) (sip.events=3Dwinfo))
>       (| (language=3Den) (language=3Dde))
>       (sip.description=3D"PC")
>       (sip.newparam=3DTRUE)
>       (rangeparam=3D-4..5125/1000))
>=20
> this declaration can be turned into this representation:
>=20
>    =
Accept-Contact:*;mobility=3D"fixed";events=3D"!presence,winfo";language=3D=
"en,de"
>     ;description=3D"<PC>";+sip.newparam;+rangeparam=3D"#-4:+5.125"
 >
> (I reversed the order as it was in the document)
>=20
> Here the events are negated and that is indicated by the exclamation =
mark at the beginning of the event-tag-value. Boolean type tags are =
turned into flag parameters with no value. So then how can the very =
first example represented?

The suppression of the TRUE value for boolean feature tags is syntactic=20
sugar, and is optional. So the following are equivalent:

	audio
	audio=3D"TRUE"

So, you can represent (! (audio=3DTRUE)) as

	audio=3D"!TRUE"


> Thanks in advance, best regards:=20
>=20
> Viktor Bekesi
> Nokia Networks
> +36-20 9849-202
> =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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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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Subject: Re: [Sip] Caller-Preferences: Representing negated Boolean feature
 tags
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Viktor.Bekesi@nokia.com wrote:
> Thank You for the answer.
> Just one last question: (audio = FALSE) is equal to not including the audio tag at all?

Not exactly. When matching two feature predicates, the matching 
algorithm considers it a match if one mentions a feature tag and another 
does not. This behavior is then modified by the presence of "require" 
and "explicit" parameters on Accept-Contact. You need to read in detail 
how it works.

	Paul


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Subject: Re: [Sip] Question regarding SDP and holds
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Chris Whitaker wrote:
> In rfc3264, it states:
> 
> If the stream to be placed on hold was previously a sendrecv media
>   stream, it is placed on hold by marking it as sendonly.
> 
> So my question is this.
> If the stream being altered was previously a sendrecv media stream, and 
> has become a sendonly stream... how does one communicate that without 
> implying that the stream is on hold [ie neither sending nor receiving]?  
> To be clear within the existing recommended proceedures for communcating 
> a hold, I am trying to understand a means of differentiating when the 
> source of media stream has stopped receiving but NOT stopped sending and 
> when it has stopped BOTH.
> 
> Secondly, if a sendrecv stream is becoming neither a sending nor 
> receiving stream [which often occurs when it goes on hold] why does the 
> rfc suggest claiming it is sendonly... would not inactive be more accurate?

I *think* the assumption is that "putting on hold" may cause something 
(music, or beeps, ...) to be sent by the holder to the holdee. So 
sendonly is used.

If the intent is truly to neither send nor receive, then it would be 
more appropriate to use inactive.

	Paul


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Date: Tue, 13 Apr 2004 10:02:41 +0300
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017978D7@esebe019.ntc.nokia.com>
Thread-Topic: Joint Interim Meeting Mon May 24 - Wed May 26
Thread-Index: AcQcABBH+m96wbjLR7CKc21Y58VQWgFJNj1gAAAYTfA=
To: <rohan@cisco.com>, <sip@ietf.org>
Cc: <dean.willis@softarmor.com>, <mankin@psg.com>, <rjsparks@nostrum.com>,
        <jon.peterson@neustar.biz>, <hardie@qualcomm.com>,
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One small correction:

Rooms will be help until 3rd May.

Thanks,
Hisham

> -----Original Message-----
> From: Khartabil Hisham (Nokia-TP-MSW/Helsinki)=20
> Sent: 13.April.2004 10:00
> To: 'ext Rohan Mahy'; 'sip@ietf.org'
> Cc: Dean Willis; Allison Mankin; Robert Sparks; John Peterson; Ted
> Hardie; Gonzalo Camarillo; Alan Johnston; Adam Roach
> Subject: RE: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> Here is the hotel information:
>=20
> Renaissance Boston Hotel Bedford
> 44 Middlesex Turnpike Bedford, MA 01730 USA
> Phone: 1 781-275-5500 Fax: 1 781-275-3042
>=20
> http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB
>=20
> There are 50 rooms booked for this event at the rate of $99=20
> per night (from Sunday 23rd until Wednesday 26th). These=20
> rooms will be held until 3th May, so please reserve your room=20
> on or before that date.
>=20
> You can reserve online or by telephone using the Group Code: NONOKA
>=20
> Also, please use the links that Rohan provided to confirm=20
> your attendance. This will aid us in determining the catering=20
> requirements. Here is the link again:
>=20
> mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[
> interim]yes
>=20
> Thanks,
> Hisham
>=20
> > -----Original Message-----
> > From: ext Rohan Mahy [mailto:rohan@cisco.com]
> > Sent: 06.April.2004 20:54
> > To: 'sip@ietf.org'
> > Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> > Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo;
> > Rohan Mahy; Alan Johnston; Adam Roach
> > Subject: Joint Interim Meeting Mon May 24 - Wed May 26
> >=20
> >=20
> > Hello All,
> >=20
> > Please mark your calendars.
> >=20
> > I'd like to announce a joint interim of the SIP, SIPPING,=20
> > SIMPLE, and =20
> > XCON WGs.
> > Nokia has generously agreed to host the event either at their=20
> > offices =20
> > near Boston or in a neighboring hotel during the week of May 24.
> >=20
> > Please reserve the following dates for the WGs that you are=20
> > interested =20
> > in:
> > Monday May 24 SIP and SIPPING  tentatively 9 am start
> > Tuesday May 25 SIMPLE   All day
> > Wednesday May 26 XCON  tentatively finish at 4pm so folks=20
> can catch =20
> > flights.
> > (the actual agenda is subject to change, including the dates=20
> > of various =20
> > WG meetings).
> >=20
> > There will be no fee, however please let Hisham and myself=20
> > know if you =20
> > are planning to attend by clicking on the Yes or Maybe links below:
> >=20
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
> >=20
> > Wireless will be provided. As usual, I will be making=20
> > T-shirts.  If you =20
> > are interested in buying one, please send your shirt size to me.
> >=20
> > More details on the agenda and hotel accommodations will be=20
> > forthcoming =20
> > shortly.
> >=20
> > thanks,
> > -rohan
> >=20
> >=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



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Date: Tue, 13 Apr 2004 09:59:55 +0300
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Thread-Topic: Joint Interim Meeting Mon May 24 - Wed May 26
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To: <rohan@cisco.com>, <sip@ietf.org>
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Here is the hotel information:

Renaissance Boston Hotel Bedford
44 Middlesex Turnpike Bedford, MA 01730 USA
Phone: 1 781-275-5500 Fax: 1 781-275-3042

http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB

There are 50 rooms booked for this event at the rate of $99 per night =
(from Sunday 23rd until Wednesday 26th). These rooms will be held until =
3th May, so please reserve your room on or before that date.

You can reserve online or by telephone using the Group Code: NONOKA

Also, please use the links that Rohan provided to confirm your =
attendance. This will aid us in determining the catering requirements. =
Here is the link again:

mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim=
]yes

Thanks,
Hisham

> -----Original Message-----
> From: ext Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 06.April.2004 20:54
> To: 'sip@ietf.org'
> Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo;
> Rohan Mahy; Alan Johnston; Adam Roach
> Subject: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> Hello All,
>=20
> Please mark your calendars.
>=20
> I'd like to announce a joint interim of the SIP, SIPPING,=20
> SIMPLE, and =20
> XCON WGs.
> Nokia has generously agreed to host the event either at their=20
> offices =20
> near Boston or in a neighboring hotel during the week of May 24.
>=20
> Please reserve the following dates for the WGs that you are=20
> interested =20
> in:
> Monday May 24 SIP and SIPPING  tentatively 9 am start
> Tuesday May 25 SIMPLE   All day
> Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch =20
> flights.
> (the actual agenda is subject to change, including the dates=20
> of various =20
> WG meetings).
>=20
> There will be no fee, however please let Hisham and myself=20
> know if you =20
> are planning to attend by clicking on the Yes or Maybe links below:
>=20
> mailto:rohan@cisco.com?=20
> Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> mailto:rohan@cisco.com?=20
> Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
>=20
> Wireless will be provided. As usual, I will be making=20
> T-shirts.  If you =20
> are interested in buying one, please send your shirt size to me.
>=20
> More details on the agenda and hotel accommodations will be=20
> forthcoming =20
> shortly.
>=20
> thanks,
> -rohan
>=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



From exim@www1.ietf.org  Tue Apr 13 09:01:56 2004
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Subject: RE: [Sip] RE: Joint Interim Meeting Mon May 24 - Wed May 26
Date: Tue, 13 Apr 2004 15:57:45 +0300
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017978DE@esebe019.ntc.nokia.com>
Thread-Topic: Joint Interim Meeting Mon May 24 - Wed May 26
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It was pointed out to me that the Group Code for booking was incorrect. =
Here is the correct Group Code: NOKNOKA.

Apologies for the inconvenience.
Hisham

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of ext
> hisham.khartabil@nokia.com
> Sent: 13.April.2004 10:00
> To: rohan@cisco.com; sip@ietf.org
> Cc: dean.willis@softarmor.com; mankin@psg.com; rjsparks@nostrum.com;
> jon.peterson@neustar.biz; hardie@qualcomm.com;
> Gonzalo.Camarillo@ericsson.com; alan.johnston@wcom.com;
> adam@dynamicsoft.com
> Subject: [Sip] RE: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> Here is the hotel information:
>=20
> Renaissance Boston Hotel Bedford
> 44 Middlesex Turnpike Bedford, MA 01730 USA
> Phone: 1 781-275-5500 Fax: 1 781-275-3042
>=20
> http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB
>=20
> There are 50 rooms booked for this event at the rate of $99=20
> per night (from Sunday 23rd until Wednesday 26th). These=20
> rooms will be held until 3th May, so please reserve your room=20
> on or before that date.
>=20
> You can reserve online or by telephone using the Group Code: NONOKA
>=20
> Also, please use the links that Rohan provided to confirm=20
> your attendance. This will aid us in determining the catering=20
> requirements. Here is the link again:
>=20
> mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[
> interim]yes
>=20
> Thanks,
> Hisham
>=20
> > -----Original Message-----
> > From: ext Rohan Mahy [mailto:rohan@cisco.com]
> > Sent: 06.April.2004 20:54
> > To: 'sip@ietf.org'
> > Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> > Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo;
> > Rohan Mahy; Alan Johnston; Adam Roach
> > Subject: Joint Interim Meeting Mon May 24 - Wed May 26
> >=20
> >=20
> > Hello All,
> >=20
> > Please mark your calendars.
> >=20
> > I'd like to announce a joint interim of the SIP, SIPPING,=20
> > SIMPLE, and =20
> > XCON WGs.
> > Nokia has generously agreed to host the event either at their=20
> > offices =20
> > near Boston or in a neighboring hotel during the week of May 24.
> >=20
> > Please reserve the following dates for the WGs that you are=20
> > interested =20
> > in:
> > Monday May 24 SIP and SIPPING  tentatively 9 am start
> > Tuesday May 25 SIMPLE   All day
> > Wednesday May 26 XCON  tentatively finish at 4pm so folks=20
> can catch =20
> > flights.
> > (the actual agenda is subject to change, including the dates=20
> > of various =20
> > WG meetings).
> >=20
> > There will be no fee, however please let Hisham and myself=20
> > know if you =20
> > are planning to attend by clicking on the Yes or Maybe links below:
> >=20
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
> >=20
> > Wireless will be provided. As usual, I will be making=20
> > T-shirts.  If you =20
> > are interested in buying one, please send your shirt size to me.
> >=20
> > More details on the agenda and hotel accommodations will be=20
> > forthcoming =20
> > shortly.
> >=20
> > thanks,
> > -rohan
> >=20
> >=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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Subject: [Sip] Presence of two Via headers in a response
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Hi all,

In RFC 3261, it is written (section 8.1.3.3 for UAC Core procedures) : =
"If more than one Via header field value is present in a response, the =
UAC SHOULD discard the message."

However if the UAC Core layer discards the response and if this response =
was the only one sent (for example a 200 OK to an UPDATE), then this =
leads to a situation when the transaction is terminated (the transaction =
layer received the response and passed it to the UA Core) but the UA =
Core is still waiting for a response (it discarded the 200 OK because of =
the presence of two Via header field values and so is still waiting for =
the response).

What are the procedures that tells the UAC when to stop waiting for the =
response ? Should there be a Timer (not described in RFC 3261) that =
tells the UAC when to CANCEL a request when no response is received ?

Additionally, what shall be the procedures when receiving more than one =
To, From, CSeq, Call-ID or Max-Forward header field values?

Thanks for your clarifications,

S=E9bastien Prouvost
France T=E9l=E9com R&D


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

<P><FONT SIZE=3D2 FACE=3D"Courier New">In RFC 3261, it is written =
(section 8.1.3.3 for UAC Core procedures) : &quot;If more than one Via =
header field value is present in a response, the UAC SHOULD discard the =
message.&quot;</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">However if the UAC Core layer =
discards the response and if this response was the only one sent (for =
example a 200 OK to an UPDATE), then this leads to a situation when the =
transaction is terminated (the transaction layer received the response =
and passed it to the UA Core) but the UA Core is still waiting for a =
response (it discarded the 200 OK because of the presence of two Via =
header field values and so is still waiting for the =
response).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">What are the procedures that =
tells the UAC when to stop waiting for the response ? Should there be a =
Timer (not described in RFC 3261) that tells the UAC when to CANCEL a =
request when no response is received ?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Additionally, what shall be the =
procedures when receiving more than one To, From, CSeq, Call-ID or =
Max-Forward header field values?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Thanks for your =
clarifications,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">S=E9bastien Prouvost</FONT>

<BR><FONT SIZE=3D2 FACE=3D"Courier New">France T=E9l=E9com =
R&amp;D</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
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From exim@www1.ietf.org  Wed Apr 14 11:11:54 2004
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Thread-Topic: Start of maintenance activity on VoIP test specification at ETSI
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From: =?iso-8859-1?Q?Fran=E7ois_Fischer?= <Francois.Fischer@etsi.org>
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Subject: [Sip] Start of maintenance activity on VoIP test specification at ETSI
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Dear all,

I would like to inform you that ETSI has just started a maintenance =
activity on the Test specification for: SIP, H.225 and H.248.

In order to improve as much as possible these Test Specifications, ETSI =
needs to collect feedback and comments on the current version.

The release of these Test Specifications will be produced as following =
(for SIP, H.225 and H.248):
PICS : 11 June 2004
TSS&TP : 23 July 2004
ATS&PIXIT : 17 September 2004

We would appreciate you to look at these test specification releases and =
to make comments latest 4 weeks before the corresponding dates above.

Please consider the Test specification as a critical part of the quality =
process to ensure Interoperability and reliability of equipments.

Furthermore, test specifications, when used by the test labs, operator =
and vendors will give confidence in VoIP and Internet Telephony =
protocols.

For more information on Testing of VoIP Protocols, and TTCN3, please =
visit:
http://www.etsi.org/ptcc/home.htm

You will also find the current release of the Test Specification in the =
following links:
Links to H.225:
PICS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00095-1/v1.0.0/
TSS&TP: http://docbox.etsi.org/MTS/MTS/07-Drafts/00095-2/v1.0.0/
ATS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00095-3/

Links to H.248:
PICS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00096-1/v1.0.0/
TSS&TP: http://docbox.etsi.org/MTS/MTS/07-Drafts/00096-2/
ATS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00096-3/

Links to SIP:
PICS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00097-1/v1.0.0/
TSS&TP: http://docbox.etsi.org/MTS/MTS/07-Drafts/00097-2/v1.0.0/
ATS: http://docbox.etsi.org/MTS/MTS/07-Drafts/00097-3/

In order to transmit your comments please use the dedicated Email =
exploder list:
H.323 : MTS-IPT-H323
H.248 : MTS-IPT-H248
SIP : MTS-IPT-SIP.

You will subscribe these lists independantly by sending an email to: =
listserv@list.etsi.org with the following command in the message body:
subscribe "listname" (firstname lastname)

eg subcribe MTS-IPT-SIP (firstname lastname)

Please contribute in maintening these specifications, which enable VoIP =
actors to ensure the right level of quality for VoIP and Internet =
Telephony items.

Many thanks in advance.


Francois FISCHER - STF expert
*************************************************
ETSI
Einstein Building - 650 route des Lucioles
F-06560 Sophia Antipolis cedex, FRANCE
*************************************************
Tel: +33 (0)4 92944330 - Fax: +33 (0)4 92385244
Email: francois.fischer@etsi.org

_______________________________________________
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: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Subject: [Sip] History-Info status - Security open issue
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Hi all,

There are still two open issues around History-Info
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-02.txt )
that require list discussion: privacy and security.  I'll post some details
on the privacy proposal in a separate posting so that I can more easily
track the responses.  

On the security, based upon some of the discussion in the e2m, m2e, policy
adhoc (per my recollection and the notes at:
http://softarmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc-ie
tf59.txt  ) there does not seem to be WG consensus that History-Info really
requires m2e (i.e. TLS may be sufficient).  I do agree that if there was an
m2e (and m2m) security mechanism in place for SIP at this point in time,
that the use of it for History-Info would add value and actually make
History-Info itself more useful to the end applications. 

What I would like to propose is that the draft reference the context of some
of these current works in progress in this area as informational and
possibly applicable to History-Info.  And, of course, define TLS as the
normative and mandatory security solution for History-Info. This results in
the security for History-Info initially being as robust as that of the other
headers added by Intermediaries (Via, etc.), per RFC 3261 and in the future,
could be made more robust.  In the end, it is up to the applications making
use of History-Info to choose the security solution that best meets its
requirements and if there is a m2m and m2e solution developed in the future,
applications can certainly take advantage of those.  

If there's general agreement on this approach, I believe this only requires
some minor rewording in section 3.2. The remaining work to complete the
draft would be updating and cleaning up the examples. 

[I realize that this will likely start a long thread on the more general
topic of m2e, which does need discussion, but if folks could, please,
upfront in their postings say whether they believe History-Info can and
should be progressed without that solution fully specified].

Regards,
Mary
mary.barnes@nortelnetworks.com



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

<P><FONT SIZE=3D2>There are still two open issues around History-Info =
(<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-=
02.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip-his=
tory-info-02.txt</A> ) that require list discussion: privacy and =
security.&nbsp; I'll post some details on the privacy proposal in a =
separate posting so that I can more easily track the responses.&nbsp; =
</FONT></P>

<P><FONT SIZE=3D2>On the security, based upon some of the discussion in =
the e2m, m2e, policy adhoc (per my recollection and the notes at: <A =
HREF=3D"http://softarmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipp=
ing-adhoc-ietf59.txt" =
TARGET=3D"_blank">http://softarmor.com/sipping/meets/ietf59/notes/notes-=
cyrus-sipping-adhoc-ietf59.txt</A>&nbsp; ) there does not seem to be WG =
consensus that History-Info really requires m2e (i.e. TLS may be =
sufficient).&nbsp; I do agree that if there was an m2e (and m2m) =
security mechanism in place for SIP at this point in time, that the use =
of it for History-Info would add value and actually make History-Info =
itself more useful to the end applications. </FONT></P>

<P><FONT SIZE=3D2>What I would like to propose is that the draft =
reference the context of some of these current works in progress in =
this area as informational and possibly applicable to =
History-Info.&nbsp; And, of course, define TLS as the normative and =
mandatory security solution for History-Info. This results in the =
security for History-Info initially being as robust as that of the =
other headers added by Intermediaries (Via, etc.), per RFC 3261 and in =
the future, could be made more robust.&nbsp; In the end, it is up to =
the applications making use of History-Info to choose the security =
solution that best meets its requirements and if there is a m2m and m2e =
solution developed in the future, applications can certainly take =
advantage of those.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>If there's general agreement on this approach, I =
believe this only requires some minor rewording in section 3.2. The =
remaining work to complete the draft would be updating and cleaning up =
the examples. </FONT></P>

<P><FONT SIZE=3D2>[I realize that this will likely start a long thread =
on the more general topic of m2e, which does need discussion, but if =
folks could, please, upfront in their postings say whether they believe =
History-Info can and should be progressed without that solution fully =
specified].</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary</FONT>
<BR><FONT SIZE=3D2>mary.barnes@nortelnetworks.com</FONT>
</P>
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
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Per my posting on the status of History-Info, Privacy is one of the
remaining open issues.  The proposal currently in the draft is quite simple:

If the Privacy header has a priv-value of "Session" or "Header" privacy, the
History-Info  SHOULD not be included in the Request (or subsequent
responses).  

This is effectively applying the fundamental recommendation in RFC3323 that
any optional information that can divulge information about the user(s)
should not be included in the requests.  

A proposal was made that there are scenarios (e.g. where local policy limits
the applicability of History-Info within a specific domain) where you would
want to capture the History-Info but not send it beyond certain
intermediaries or domains.  To support this, it was proposed that we could
extend the Privacy header defined in RFC 3323 with tags for History-Info
something like the following:

priv-value = "history" / "history-index"  

The "history" priv-value would mean that all the History-Info entries should
be removed.  The "history-index" would be used to indicate that the specific
entry is to be removed and would need to have the Privacy header escaped in
the URI so that it can be associated with a specific History-Info entry (or
we need to add another field to History-Info for this).    

So, I think we have two options here:
1) Extend History-Info to have an optional privacy field along the lines of
what's detailed above.  

2) Defer the definition of the privacy field to a separate draft, since it
appears to be primarily limited to scenarios with certain trust models.  

So, if folks could just take a bit of time and consider their preference, I
would like to get the draft updated soon and closed off prior to the Interim
meeting.

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

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<TITLE>History-Info Privacy Issue</TITLE>
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<P><FONT SIZE=3D2>Per my posting on the status of History-Info, Privacy =
is one of the remaining open issues.&nbsp; The proposal currently in =
the draft is quite simple: </FONT></P>

<P><FONT SIZE=3D2>If the Privacy header has a priv-value of =
&quot;Session&quot; or &quot;Header&quot; privacy, the =
History-Info&nbsp; SHOULD not be included in the Request (or subsequent =
responses).&nbsp; </FONT></P>

<P><FONT SIZE=3D2>This is effectively applying the fundamental =
recommendation in RFC3323 that any optional information that can =
divulge information about the user(s) should not be included in the =
requests.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>A proposal was made that there are scenarios (e.g. =
where local policy limits the applicability of History-Info within a =
specific domain) where you would want to capture the History-Info but =
not send it beyond certain intermediaries or domains.&nbsp; To support =
this, it was proposed that we could extend the Privacy header defined =
in RFC 3323 with tags for History-Info something like the =
following:</FONT></P>

<P><FONT SIZE=3D2>priv-value =3D &quot;history&quot; / =
&quot;history-index&quot;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>The &quot;history&quot; priv-value would mean that =
all the History-Info entries should be removed.&nbsp; The =
&quot;history-index&quot; would be used to indicate that the specific =
entry is to be removed and would need to have the Privacy header =
escaped in the URI so that it can be associated with a specific =
History-Info entry (or we need to add another field to History-Info for =
this).&nbsp;&nbsp;&nbsp; </FONT></P>

<P><FONT SIZE=3D2>So, I think we have two options here:</FONT>
<BR><FONT SIZE=3D2>1) Extend History-Info to have an optional privacy =
field along the lines of what's detailed above.&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>2) Defer the definition of the privacy field to a =
separate draft, since it appears to be primarily limited to scenarios =
with certain trust models.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>So, if folks could just take a bit of time and =
consider their preference, I would like to get the draft updated soon =
and closed off prior to the Interim meeting.</FONT></P>

<P><FONT SIZE=3D2>Regards, </FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>mary.barnes@nortelnetworks.com</FONT>
</P>

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From: Fredrik Thulin <ft@it.su.se>
To: Alex Bligh <alex@alex.org.uk>
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Subject: [Sip] unanswered: RFC3261 & authenticating PROXIES to other proxies & UAS's
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On Wednesday 07 April 2004 17.41, Alex Bligh wrote:
> --On 06 April 2004 18:39 +0200 Fredrik Thulin <ft@it.su.se> wrote:
> > There is wording in section 22.3 saying proxys MUST NOT add things to
> > authorization headers, and that all 407 responses MUST be forwarded
> > upstreams. That is a pity, since I think otherwise the following could
> > solve  the problem :
> >
> > P1 could, upon receiving the 407 from P2, start a new transaction with a
> > different branch parameter and added authentication credentials. The
> > request  could then be re-submitted to P2.
>
> If I read things right, the branch parameter is going to be different
> /anyway/ for a proxy with loop detection:
>
> 16.6 Request Forwarding [Section 8]
>
> ...
>          If a proxy wishes to detect loops, the "branch" parameter it
>          supplies MUST depend on all information affecting processing of
>          a request, including the incoming Request-URI and any header
>          fields affecting the request's admission or routing.  This is
>          necessary to distinguish looped requests from requests whose
>          routing parameters have changed before returning to this
>          server.
>
> On this basis, if the proxy retries the INVITE with the same CSeq,
> what is the problem? IE how is this any different from spiraling
> (which is allowed)?

I think you have a very good point here. Now that people should be well back 
from their easter vacation I would like to ask again that someone explains 
the reason for explicitly forbidding proxys to respond to 407 challenges.

/Fredrik


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From exim@www1.ietf.org  Mon Apr 19 15:51:56 2004
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To: Fredrik Thulin <ft@it.su.se>
Cc: Alex Bligh <alex@alex.org.uk>, sip@ietf.org
Subject: Re: [Sip] unanswered: RFC3261 & authenticating PROXIES to other proxies
 & UAS's
References: <200404190815.24768.ft@it.su.se>
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Hi!

I think I found another problem - if the proxy inserts the 
Authentication header, the authentication user will likely be different 
to the user in the From: header. Such messages typically will be refused 
by a SIP proxy as this allows to fake the sender id.

regards,
Klaus

Fredrik Thulin wrote:
> On Wednesday 07 April 2004 17.41, Alex Bligh wrote:
> 
>>--On 06 April 2004 18:39 +0200 Fredrik Thulin <ft@it.su.se> wrote:
>>
>>>There is wording in section 22.3 saying proxys MUST NOT add things to
>>>authorization headers, and that all 407 responses MUST be forwarded
>>>upstreams. That is a pity, since I think otherwise the following could
>>>solve  the problem :
>>>
>>>P1 could, upon receiving the 407 from P2, start a new transaction with a
>>>different branch parameter and added authentication credentials. The
>>>request  could then be re-submitted to P2.
>>
>>If I read things right, the branch parameter is going to be different
>>/anyway/ for a proxy with loop detection:
>>
>>16.6 Request Forwarding [Section 8]
>>
>>...
>>         If a proxy wishes to detect loops, the "branch" parameter it
>>         supplies MUST depend on all information affecting processing of
>>         a request, including the incoming Request-URI and any header
>>         fields affecting the request's admission or routing.  This is
>>         necessary to distinguish looped requests from requests whose
>>         routing parameters have changed before returning to this
>>         server.
>>
>>On this basis, if the proxy retries the INVITE with the same CSeq,
>>what is the problem? IE how is this any different from spiraling
>>(which is allowed)?
> 
> 
> I think you have a very good point here. Now that people should be well back 
> from their easter vacation I would like to ask again that someone explains 
> the reason for explicitly forbidding proxys to respond to 407 challenges.
> 
> /Fredrik
> 
> 
> _______________________________________________
> 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, 19 Apr 2004 15:38:15 -0400
From: "David R. Oran" <oran@cisco.com>
To: Mary Barnes <mary.barnes@nortelnetworks.com>, sip@ietf.org
Subject: Re: [Sip] History-Info status - Security open issue
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I think it's fine to allow history-info to progress before we have e2m=20
security nailed. Think of how many hacks we can deprecate (which themselves =

present the same security concerns) if we finally get this beastie=20
last-called.

so, put me down in the "yes" column.

Dave.

--On Friday, April 16, 2004 11:04 AM -0500 Mary Barnes=20
<mary.barnes@nortelnetworks.com> wrote:

>
> Hi all,
>
> There are still two open issues around History-Info
> (http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-02.txt )
> that require list discussion: privacy and security.  I'll post some
> details on the privacy proposal in a separate posting so that I can more
> easily track the responses.
>
> On the security, based upon some of the discussion in the e2m, m2e,
> policy adhoc (per my recollection and the notes at:
> http://softarmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc
> -ietf59.txt  ) there does not seem to be WG consensus that History-Info
> really requires m2e (i.e. TLS may be sufficient).  I do agree that if
> there was an m2e (and m2m) security mechanism in place for SIP at this
> point in time, that the use of it for History-Info would add value and
> actually make History-Info itself more useful to the end applications.
>
> What I would like to propose is that the draft reference the context of
> some of these current works in progress in this area as informational and
> possibly applicable to History-Info.  And, of course, define TLS as the
> normative and mandatory security solution for History-Info. This results
> in the security for History-Info initially being as robust as that of the
> other headers added by Intermediaries (Via, etc.), per RFC 3261 and in
> the future, could be made more robust.  In the end, it is up to the
> applications making use of History-Info to choose the security solution
> that best meets its requirements and if there is a m2m and m2e solution
> developed in the future, applications can certainly take advantage of
> those.
>
> If there's general agreement on this approach, I believe this only
> requires some minor rewording in section 3.2. The remaining work to
> complete the draft would be updating and cleaning up the examples.
>
> [I realize that this will likely start a long thread on the more general
> topic of m2e, which does need discussion, but if folks could, please,
> upfront in their postings say whether they believe History-Info can and
> should be progressed without that solution fully specified].
>
> Regards,
> Mary
> mary.barnes@nortelnetworks.com




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_______________________________________________
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From exim@www1.ietf.org  Mon Apr 19 16:12:48 2004
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Date: Mon, 19 Apr 2004 15:41:04 -0400
From: David Oran <oran@cisco.com>
To: Mary Barnes <mary.barnes@nortelnetworks.com>, sip@ietf.org
Subject: Re: [Sip] History-Info Privacy Issue
Message-ID: <2147483647.1082389264@[10.32.245.156]>
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Either resolution is fine with me. dave.

--On Friday, April 16, 2004 11:05 AM -0500 Mary Barnes 
<mary.barnes@nortelnetworks.com> wrote:

>
> Per my posting on the status of History-Info, Privacy is one of the
> remaining open issues.  The proposal currently in the draft is quite
> simple:
>
> If the Privacy header has a priv-value of "Session" or "Header" privacy,
> the History-Info  SHOULD not be included in the Request (or subsequent
> responses).
>
> This is effectively applying the fundamental recommendation in RFC3323
> that any optional information that can divulge information about the
> user(s) should not be included in the requests.
>
> A proposal was made that there are scenarios (e.g. where local policy
> limits the applicability of History-Info within a specific domain) where
> you would want to capture the History-Info but not send it beyond certain
> intermediaries or domains.  To support this, it was proposed that we
> could extend the Privacy header defined in RFC 3323 with tags for
> History-Info something like the following:
>
> priv-value = "history" / "history-index"
>
> The "history" priv-value would mean that all the History-Info entries
> should be removed.  The "history-index" would be used to indicate that
> the specific entry is to be removed and would need to have the Privacy
> header escaped in the URI so that it can be associated with a specific
> History-Info entry (or we need to add another field to History-Info for
> this).
>
> So, I think we have two options here:
> 1) Extend History-Info to have an optional privacy field along the lines
> of what's detailed above.
>
> 2) Defer the definition of the privacy field to a separate draft, since
> it appears to be primarily limited to scenarios with certain trust
> models.
>
> So, if folks could just take a bit of time and consider their preference,
> I would like to get the draft updated soon and closed off prior to the
> Interim meeting.
>
> Regards,
> Mary H. Barnes
> mary.barnes@nortelnetworks.com





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From exim@www1.ietf.org  Mon Apr 19 16:30:41 2004
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Date: Mon, 19 Apr 2004 21:00:32 +0100
From: Alex Bligh <alex@alex.org.uk>
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To: Klaus Darilion <klaus.mailinglists@pernau.at>,
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Subject: Re: [Sip] unanswered: RFC3261 & authenticating PROXIES to other
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--On 19 April 2004 21:31 +0200 Klaus Darilion 
<klaus.mailinglists@pernau.at> wrote:

> I think I found another problem - if the proxy inserts the Authentication
> header, the authentication user will likely be different to the user in
> the From: header. Such messages typically will be refused by a SIP proxy
> as this allows to fake the sender id.

That should not create a problem. The original proxy (the one inserting
the header) will presumably have already authorized the user to /its/
satisfaction if it is going to then give its own credentials to the
downstream proxy, i.e. in
   UAC ------> P1 ------> P2 ------> UAS
P1 can authenticate UAC against its own database; only if it is
authenticated will it in turn respond to an authentication request
from either P2 or UAS.

It is /desirable/ that the authentication user is different from the From:
user. This allows P1 to identify /itself/ (in the authentication user). The
idea of the scheme is to prevent P2, UAC from having to know credentials
about UA's such as UAC.

Whether the SIP Proxy P2 or the UAS refuses packets with an authentication
user different from the From: user is a matter of local policy, and is
explicitly not mandated in the standard. In SER, for example, you
have to explicitly require that it checks this. It is quite legal for
the auth user to differ from the From: user.

All P2/UAS wants to know is that the call has been authorized **BY P1**.
Whether this is the "remote media gateway in different administrative
zone" problem, or the "I wish to authorize this SIP call to be diverted
at my expense to my cellphone" problem...

Alex

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Alex Bligh wrote:
> 
...
> Whether the SIP Proxy P2 or the UAS refuses packets with an authentication
> user different from the From: user is a matter of local policy, and is
> explicitly not mandated in the standard. In SER, for example, you
> have to explicitly require that it checks this. It is quite legal for
> the auth user to differ from the From: user.
I do this check - but only for local users. So if P2 is the proxy of any 
  termination provider, the From: will be an external domain for P2 and 
no check is necessary.

Short answer: you are right - i shot too fast.

I'm currently thinking about a really dirty hack for this:
before forwarding the request in P1, the request will be handled to an 
externel script (exec module in ser). The script takes the request, 
decreases cseq by 1, sends the request to P2, catches the response, 
filters realm and nonce, calculates the hash and gives the new 
Authentication header back to ser (exec must be enhanced to add 
headers). Then ser will forward the request, which should now include 
proper credentials.

Yes of course this is terrible and wont scale. :-)

klaus


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To: Klaus Darilion <klaus.mailinglists@pernau.at>
Cc: Fredrik Thulin <ft@it.su.se>, sip@ietf.org, Alex Bligh <alex@alex.org.uk>
Subject: Re: [Sip] unanswered: RFC3261 & authenticating PROXIES to other
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--On 19 April 2004 22:32 +0200 Klaus Darilion 
<klaus.mailinglists@pernau.at> wrote:

> I'm currently thinking about a really dirty hack for this:
...
> Then ser will forward the request, which should now include proper
> credentials.

It doesn't actually look that hard to hack it in ser. But I know I will
(rightly) get far more support from the iptel.org folks if there is some
support for this "not being against the standard" or rather "a useful
modification to the standard", hence my asking the question. I am
sufficiently confident/pigheaded to think "it will work" anyway.

[ your hack decrements cseq by one - I think you will run into the
  possibility with a collision with a previous message, and that the
  proxy etc. may thus ignore it thinking it's a retransmission - you
  would be better to use the same cseq but insert a bogus branch
  value; but even better, do it properly :-) ]

Alex

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From exim@www1.ietf.org  Mon Apr 19 18:19:23 2004
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From: Klaus Darilion <klaus.mailinglists@pernau.at>
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To: Alex Bligh <alex@alex.org.uk>
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Subject: Re: [Sip] unanswered: RFC3261 & authenticating PROXIES to other proxies
 & UAS's
References: <200404190815.24768.ft@it.su.se> <4084290A.9020602@pernau.at> <4156953332.1082408432@[192.168.100.5]> <40843756.8070707@pernau.at> <4161870954.1082413350@[192.168.100.5]>
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I just come around another problem:

The reason for me to let the proxy doing the authentication is for 
example the following scenario:

A small company (@smallcomp.com) has its own sip proxy (P1). Local and 
IP calls will work fine. Now, the company wants to make also PSTN calls, 
and gets a single account by any termination provider (vonage, sipgate, 
nikotel...) -> P2. So, their proxy P1 should forward PSTN calls to P2 
and add credentials.

And here is my problem - P2 will verify the From: header. P2 identifies 
an external user (@smallcomp.com != @nikotel.com) and refuses the 
request as externel users are not allowed to access the gateway, even if 
they would send proper credentials. Only local users will be asked for 
credentials.

So, I can't use such a workaround anyway.

and more generic: how will P2 know if the request from P1 should be 
challenged or handled as incoming call (without challenging)?

klaus

Alex Bligh wrote:
> 
> 
> --On 19 April 2004 22:32 +0200 Klaus Darilion 
> <klaus.mailinglists@pernau.at> wrote:
> 
>> I'm currently thinking about a really dirty hack for this:
> 
> ...
> 
>> Then ser will forward the request, which should now include proper
>> credentials.
> 
> 
> It doesn't actually look that hard to hack it in ser. But I know I will
> (rightly) get far more support from the iptel.org folks if there is some
> support for this "not being against the standard" or rather "a useful
> modification to the standard", hence my asking the question. I am
> sufficiently confident/pigheaded to think "it will work" anyway.
> 
> [ your hack decrements cseq by one - I think you will run into the
>  possibility with a collision with a previous message, and that the
>  proxy etc. may thus ignore it thinking it's a retransmission - you
>  would be better to use the same cseq but insert a bogus branch
>  value; but even better, do it properly :-) ]
> 
> Alex
> 
> _______________________________________________
> 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] RFC3261 and authenticating PROXIES to other proxies & U
	AS - 2 possible solutions
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I have some (lengthy) comments on this.

First of all, "authenticating that a call has been transmitted by a proxy
that the headers claimed transmitted it" is an aspect of a relatively
familiar SIP problem called the "request history" problem. It is difficult
to discuss this issue, or any solutions to it, outside of the long
back-story of request history and middle-to-middle security. Please see Mary
Barnes' work on the subject, especially her requirements and solutions
drafts (some references are at the end of this mail).

For the very simple SIP trapezoid case you describe below, there is of
course a solution provided in RFC3261, specifically in 26.3.2.2: P1 and P2
would use mutual TLS to authenticate one another. In that fashion, P2 would
have the necessary assurance that "the call came from P1", as your
requirements below state. P1 is happy if the call came from the UAC, P2 is
happy if it came from P1, and the UAS is happy if it came from P2 - we call
all of this collective happiness "transitive trust". And unlike your
solutions, it does not require that P1 and P2 exchange any keying
information with one another. So why isn't 26.3.2.2 sufficient to solve all
request history problems? Because sometimes transitive trust is
insufficient. Request history also takes into account cases where the UAS,
for example, would want to have a direct cryptographic assurance that P1
handled the call, rather than just being content to know that P2 forwarded
it.

So, if you're content with the simple transitive trust case, there's a
solution for you already in RFC3261 - transitive trust and mutual TLS
between P1 and P2. If you require something stronger than transitive trust,
we've landed ourselves in request history / m2m territory.

Now, specifically speaking to your solutions: the first one, as this thread
noted earlier, suffers from a transaction-matching problem. In a nutshell,
the problem is this: if the proxy server resubmits the INVITE (adding a
Proxy-Authorization) with a new CSeq, then the UAC will reject the final
response from the UAS since it is out of sequence (vide RFC3261 22.3). So
why shouldn't P1 just resubmit the INVITE with the old CSeq and a
Proxy-Authorization header? Well, for the sake of argument, let's say it
could (although I remain a little nervous that implementation snafus with P2
matching nonces/transactions/branches/CSeqs could arise, since RFC3261 says
that the resubmitted INVITE will arrive on a new CSeq).

This approach suffers from the general problem of conflating the semantics
of two distinct operations. Suppose that P1 resubmits the INVITE with
credentials that are not accepted by P2, for whatever reason? How is the UAC
going to understand that situation when the call fails? Also, shouldn't the
UAC have its own opportunity to supply credentials to P2? If the user of the
UAC has their own account/relationship with P2, they might want to rely on
that rather than on allowing P1 to assert its own credentials - P1 is
essentially interposing in the challenge. Essentially, by allowing P1 to
consume the 407, you are proposing a change to the security semantics of the
Digest mechanism. If I'm operating a proxy server, and I send a 407, I
expect that the UAC will handle it and resubmit. I can imagine various
scenarios in which I'd be unhappy if that challenge were intercepted and
answered by a proxy server. If we did want to do something like this, I
imagine we'd want a specific mechanism (new response code, or something)
that makes it unambiguous that P2 intends to challenge P1 - see the m2m
work.

Moreover, there is a security concern here. The HTTP Digest authentication
mechanism uses the concept of 'realms' (as has previously been discussed in
this thread). In order to prevent a malicious proxy server from
impersonating a realm (i.e., P3 spoofs, with its own nonces and so on, a
challenge from P2 asserting the realm of P2, in an attempt to capture or
interrupt authentication for P2's realm), realms must be correlated with
certificate-based security. In fact, the recommendations in RFC3261
specifically state that the Digest realm should be identical to the
subjectAltName in the certificate of the issuing proxy server - if they are
not, there's always a potential for a security concern (26.3.2 passim).
Accordingly, the establishment of a TLS connection (and the resulting
certificate exchange) is more or less a prerequisite for the secure use of
HTTP Digest. But of course, in your case, if certificates are exchanged, the
use of HTTP Digest is entirely superfluous - you don't really care about the
usernames/passwords, you just want to know which proxy server is forwarding
the request. You'll get that information from TLS anyway. So using HTTP
Digest without TLS is insecure, using it with TLS is superfluous.

In short, my recommendation would be just to use mutual TLS.

Regarding your second proposed solution, I think the n-squared problem of
distributing and managing shared secrets for MD5 hashes in Via headers is
intractable. For one thing, it requires you (as P1, say) to be able to
anticipate ALL of the proxy servers/UASes to which your request might be
forwarded, and to include suitable MD5 hashes for each. But the problem is,
a proxy in the position of P1 simply doesn't know where their request is
going to be forwarded. Sure, they know it is going to P2, but what if P2
forwards the request to P3 (because all of its UASes are overloaded, or
something)? How is P1 supposed to know that it needs to insert a vauth
parameter that will be verifiable by P3? Moreover, why on earth would P1
have exchanged a shared secret to every possible destination to which its
requests could be forwarded? The fact that proxy servers are not psychic
makes this extremely problematic. Moreover, since it is not a
challenge-response system, it has extreme forgeability/replay problems.
While the 'vauthexpire' parameter is obviously intended to mitigate the
replay issue, as written, it is impracticable (i.e. messages containing it
are silently discarded if vauthexpire is unsupported or invalid).

And finally, as you hint, a malicious proxy server or man-in-the-middle
could just strip this vauth parameter anyway. But of course, the only reason
to have this parameter is because you don't trust the Via headers in the
first place. And why don't you trust the Via headers? Because a malicious
proxy server or man-in-the-middle could just strip or modify Via headers. So
how exactly has the vauth parameter improved our situation? You don't trust
an intermediary to keep leave the meat of the Via headers untouched, but you
do trust them to leave vauth untouched? Mutual TLS between proxy servers
prevents interceptors between the proxies from modifying signaling, but if
the proxies are misbehaving, you're in the same boat with or without vauth.
Might as well just use SIPS and mutual TLS to protect your Vias, or do
something like request history and/or m2m.

Earlier in the request history work, there were a number of proposed
via-signing solutions along the lines of your second solution. I don't think
an MD5-based Via header parameter is a profitable line of inquiry into this
problem. I'd suggest contributing to the request history work if you're
interested in some of the possible alternatives. Today, the mature form of
that work is in:

http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-02.txt

You might also be interested inb the work on middle-to-middle security in:

http://www.ietf.org/internet-drafts/draft-barnes-sipping-sec-inserted-info-0
1.txt

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Alex Bligh [mailto:alex@alex.org.uk]
> Sent: Thursday, April 08, 2004 5:49 AM
> To: sip@ietf.org
> Cc: Alex Bligh
> Subject: [Sip] RFC3261 and authenticating PROXIES to other 
> proxies & UAS
> - 2 possible solutions
> 
> 
> I am still trying to work out how one might authenticate that a call has
> been transmitted by a proxy that the headers claimed transmitted it. SIP
> provides facilities (in protocol) for proxies and UAS's to determine that
a
> call originated at a UAC with credentials, but no manner of determining
> whether the call passed through a proxy with credentials. Whilst in some
> circumstances this can be handled with network layer or transport layer
> security, (a) this requires support on a hop by hop basis from the proxy
> downstream, (b) the overhead may be substantial, and (c) it is somewhat
> inconsistent that SIP supports this type of authentication 
> for the UAC but
> not proxies.
> 
> I've come up with 2 mechanisms to provide the support - one I posted
> earlier (which requires only a change on the proxy with 
> credentials) and
> goes into the "Despite being against RFC3261, I can't see 
> what this would
> break" category, and the second (which requires a change both 
> at the proxy
> supplying credentials and at the server authenticating them) certainly
> breaks nothing, but is less functional.
> 
> I'd appreciate comments, particularly on what the first 
> mechanism breaks.
> I (now) know that it is (today) against various bits of RFC3261. But I
> cannot see what in practice it would break. And as a couple of people
> (on and off list) have pointed out, it would be very useful (even for
> reasons I hadn't thought of).
> 
> BACKGROUND
> ==========
> 
> The (general) problem I am trying to solve looks like this:
> 
> 
>    UAC <-----/\/\/\/\/---> P1 <--/\/--> P2 <---> UAS => PSTN
>   (CPE)                 (proxy)      (proxy)  (media
>                                               gateway)
> 
> P2 may or may not be present.
> 
> Let us assume UAC and P1 are very distant, and P1 & either P2 
> or UAS are
> sufficiently distant that host security is impractical 
> without adding some
> additional layer (ipsec, tls etc.).
> 
> UAC is CPE on a site of a customer of SP1, who operates P1. 
> UAS is a media
> gateway, and P2 a (possibly) intervening proxy, operated by SP2.
> 
> Let us further assume that UAC registers to P1, and that P1 
> has already
> authenticated UAC, and that any INVITE passed through P1 has 
> already been
> authenticated as coming from UAC and been authorized by P1 
> according to
> SP1's local policy, prior to being forwarded to P2.
> 
> Let us further assume that it is the local policy of SP1 that any such
> authorized calls it passes to SP2 would be authorized by SP1. So for
> instance SP2 is a VoIP termination provider, and SP1 a VoIP service
> provider.
> 
> SP2 wishes to authenticate & authorize calls made to UAS. It 
> uses some form
> of authentication. What it in essence is trying to 
> authenticate is that the
> call actually came from P1, not that the call actually came 
> from UAC. This
> could either be done at P2, or at UAS.
> 
> I have two mechanisms of doing this
> 
> MECHANISM (AB)USING HTTP DIGEST AUTHENTICATION
> ==============================================
> 
> As per previous email, (ab)use HTTP digest authentication so
> that the proxy retries the Invite with the same CSeq but with
> a different branch parameter. This seems to me indistinguishable
> from a spiraling scenario where the proxy concerned happens to
> call the same Request-URI twice but other information in the message
> header (in this case authentication) that may be used to route the
> call has changed.
> 
> As such, I can't see what it would break, and it has the major
> advantage it works with all existing (challenging) UA's & proxies,
> and that being challenge-response, it is more secure than
> other mechanisms.
> 
> But, for the sake of argument
> 
> ALTERNATIVE MECHANISM USING vauth Via-Parameter
> ===============================================
> 
> This mechanism introduces a new Via Parameter (vauth).
> 
> A UAC or Proxy inserting a Via Header with a RFC3261 compliant branch
> parameter MAY introduce one or vauth parameter into the Via 
> header, in each
> case to identify authenticate the via header to one or more downstream
> proxies or UAS's. Such UAS's or proxies MAY use a vauth header to take
> local-policy dependent routing or access decisions. A UAC or 
> Proxy which
> does not insert a Via Header with a RFC3261 compliant branch 
> parameter MUST
> NOT insert any vauth parameters.
> 
> The vauth parameter is intended to allow a server to 
> authenticate a subset
> of the Via: header (being the sent-by entry and the branch 
> token), together
> with an opaque string, to downstream servers. Multiple vauth 
> parameters MAY
> be used in order to support multiple realms of 
> authentication. The vauth
> parameter supports extensible mechanisms of authentication, but the
> mechanism specified herein is MD5 authentication using a 
> shared secret.
> The mechanism of exchange of the shared secret and 
> authentication realm
> is outside the scope of this document.
> 
> The vauthid field specifies the realm of authentication. The 
> vauthid field
> MUST be unique across instances if the vauth parameter in a 
> single Via:
> header. Those specifying vauth fields SHOULD ensure that they 
> are globally
> unique, for example a domain name MAY be used.
> 
> The vauthopaque field is an opaque field intended to pass 
> information to
> servers in the authentication realm which MAY be used to 
> influence their
> routing decisions - for instance it might be used to specify 
> that access
> should be permitted or denied, or have an accounting function.
> 
> The vauthmeth field specifies the mechanism of 
> authentication. The only
> currently defined method of authentication is MD5. In this 
> mechanism, the
> vauthmd5data field contains a hash of the sent-by field, the branch
> parameter (which is, by RFC3261 unique across time and 
> space), the vauthid
> field, the vauthopaque field, the vauthexpire field (if any), and the
> shared secret, which should be a string of printable 
> characters no more
> than 64 octets in length. The hash is calculated using MD5 
> (RFC 1321 [35]),
> expressed in hexadecimal.
> 
> The optional vauthexpire field is a date in the SIP date 
> format (i.e. the
> preferred RFC1123 format) after which it should be assumed that the
> header is not authentic. If a vauthexpire field is inserted, 
> the server
> MUST be synchronized to an accurate time source, and the expiry time
> MUST be no shorter than 32 seconds beyond the time at which the Via:
> header is to be transmitted.
> 
> A server MAY check the authenticity of a Via: header using 
> vauth parameters
> by performing each of the following checks:
> 
> 1. Isolate the vauth field(s) corresponding to a realm for which the
>    checking server possesses a shared secret.
> 
> 2. If vauthexpire is present: if the checking server does not support
>    vauthexpire, it MUST silently discard the request. If the checking
>    server does support vauthexpire, it MUST check that the current
>    time is not beyond the expiry time, and if it is, it MUST
>    silently discard the request, else proceed.
>    The silent discards are used to ensure that excessive
>    transmission delays result only in the symptoms of packet loss,
>    rather than false authentication refusals.
> 
> 3. The server MUST recalculate the hash value and check whether
>    it corresponds to the hash value in the vauthmd5data field.
> 
> 4. The result of (3), the vauthopaque field, and the presence or
>    absence of a vauthexpire field, MAY then be used
>    to influence routing decisions
> 
> The vauth mechnanism is primarily designed to protect against spoofing
> attacks, as the shared secret will not be known. It provides 
> only limited
> protection against replay attacks, in that replays are 
> time-limited by the
> vauth-expire header. Further limitation of replay attacks can 
> be given by
> the implementor using the vauthopaque field - for instance in an
> environment where the Request-URI is known not to change 
> between the server
> inserting the Via header and the server checking it, the 
> vauthopaque field
> could be set to contain a hash of the Request-URI, which 
> could be checked
> upon receipt. Further, it should be noted that mechanism provides no
> protection against malicious or accidental modification of the message
> between transmission by the server inserting the Via: header 
> and the server
> analyzing it. It is not intended as a substitute for network 
> or transport
> layer security.
> 
> An example of a Via: header with vauth parameters follows:
> Via: SIP/2/0/UDP 4.3.2.1:5060 branch=1fe452.12;
>  
> vauth="server.example.com:5061,y,MD5=1234567890abcdef123456789
> 0abcdef";
>  
> vauth="server2.example.com:5060,,MD5=fedcba0987654321fedcba098
> 7654321:Fri, 
> 13 Oct 1998 23:50:00 GMT"
> 
> RFC3261 specifies:
> 
>     via-params        =  via-ttl / via-maddr
>                          / via-received / via-branch
>                          / via-extension
>     via-extension     =  generic-param
>     generic-param     =  token [ EQUAL gen-value ]
>     gen-value         =  token / host / quoted-string
>     token             =  1*(alphanum / "-" / "." / "!" / "%" / "*"
>                         / "_" / "+" / "`" / "'" / "~" )
> 
> This document introduces a new via-param which matches the 
> generic-param
> model:
> 
>     via-params        =  via-ttl / via-maddr
>                          / via-received / via-branch
>                          / via-vauth ; new
>                          / via-extension
>     via-vauth         =  "vauth" EQUAL vauth
>     vauth             = LTQUOT vauthid COMMA vauthopaque
>                         COMMA vauthmeth [ COLON vauthexpire ]RDQUOT
>     vauthid           = vauthstr
>     vauthopaque       = [ vauthstr ]
>     vauthexpire       = SIP-date
>     vauthstr          = 1*(alphanum / "-" / "." / "/" / ":" / "_" /
>                         "[" / "]")
>     vauthmeth         = vauthmeth-md5 / vauthmeth-ext
>     vauthmeth-ext     = vauthextension EQUAL vauthextdata
>     vauthextension    = 1*(alphanum)
>     vauthextdat       = 1*(alphanum)
>     vauthmeth-md5     = vauthmmd5 EQUAL vauthmd5data
>     vauthmmd5         = "MD5"
>     vauthmd5data      = 32LHEX
> 
> 
> Alex
> 
> _______________________________________________
> 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 exim@www1.ietf.org  Mon Apr 19 19:20:02 2004
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From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce:;
Cc: Internet Architecture Board <iab@iab.org>,
        RFC Editor <rfc-editor@rfc-editor.org>,
        sip mailing list <sip@ietf.org>, sip chair <dean.willis@softarmor.com>,
        sip chair <rohan@cisco.com>
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Date: Mon, 19 Apr 2004 18:34:42 -0400
Subject: [Sip] Protocol Action: 'S/MIME AES Requirement for SIP' to
 Proposed Standard
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The IESG has approved the following document:

- 'S/MIME AES Requirement for SIP '
   <draft-ietf-sip-smime-aes-01.txt> as a Proposed Standard

This document is the product of the Session Initiation Protocol Working Group. 

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary
 
   RFC3261 currently specifies 3DES as the required minimum ciphersuite
   for implementations of S/MIME in SIP.  This document updates the
   normative guidance of RFC3261 to require the Advanced Encryption
   Standard (AES) for S/MIME.
 
Working Group Summary
 
   The Working Group supported this document.  It was adopted immediately
   on its initial airing.  It was gated by progress on S/MIME support.
 
 Protocol Quality
 
   General S/MIME implementation for SIP has been fairly slow to progress.  
   Some prototype implementations have been tested at the SIP 
   interoperability events, without testing their cryptography to date.

The specification was reviewed for the IESG by Allison Mankin and Russ
Housley.


RFC Editor Notes 

OLD:
   S/MIME implementations MUST at a minimum support RSA as a digital
   signature algorithm, SHA1 as a digest algorithm, and AES as an
   encryption algorithm (as specified in [4].  For key wrap, S/MIME
   implementations MUST support the AES Key Wrap Algorithm ([5]).  

NEW:
   S/MIME implementations MUST at a minimum support RSA as a digital
   signature algorithm and SHA1 as a digest algorithm [ xx],  and AES as
   an encryption algorithm (as specified in [yy]).  For key transport, 
   S/MIME implementations MUST support RSA key transport as specified
   in section 4.2.1 of [xx].  

RFC Editor, replace [xx] with the citation number of a reference to RFC 3370
added to the Normative References.  Replace [yy] with the citation number
of a reference to RFC 3565 added to the Normative References.

3370 Cryptographic Message Syntax (CMS) Algorithms. R. Housley.
    August 2002.

3565 Use of the Advanced Encryption Standard (AES) Encryption
     Algorithm in Cryptographic Message Syntax (CMS). J. Schaad.
     July 2003.

****

Abstract

OLD:
  required minimum ciphersuite
NEW:
  mandatory-to-implement ciphersuite

****

Section 4

OLD:
  Triples-DES

NEW:
  Triple-DES

****

   Several places: Adjust line breaks to avoid funny
   line break placement --  Avoid  S/ <CR><LF> MIME


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To: "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: "'Alex Bligh'" <alex@alex.org.uk>, sip@ietf.org
Subject: RE: [Sip] RFC3261 and authenticating PROXIES to other proxies & U
	AS - 2 possible solutions
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Peterson, Jon writes:
 > Moreover, there is a security concern here. The HTTP Digest authentication
 > mechanism uses the concept of 'realms' (as has previously been discussed in
 > this thread). In order to prevent a malicious proxy server from
 > impersonating a realm (i.e., P3 spoofs, with its own nonces and so on, a
 > challenge from P2 asserting the realm of P2, in an attempt to capture or
 > interrupt authentication for P2's realm), realms must be correlated with
 > certificate-based security. In fact, the recommendations in RFC3261
 > specifically state that the Digest realm should be identical to the
 > subjectAltName in the certificate of the issuing proxy server - if they are
 > not, there's always a potential for a security concern (26.3.2 passim).
 > Accordingly, the establishment of a TLS connection (and the resulting
 > certificate exchange) is more or less a prerequisite for the secure use of
 > HTTP Digest. But of course, in your case, if certificates are exchanged, the
 > use of HTTP Digest is entirely superfluous - you don't really care about the
 > usernames/passwords, you just want to know which proxy server is forwarding
 > the request. You'll get that information from TLS anyway. So using HTTP
 > Digest without TLS is insecure, using it with TLS is superfluous.

   I can think of a pretty trivial example, I think,
   that breaks this assumption. Suppose that I'm at
   a public kiosk UAC and want to read my voice mail
   back at mega.corprahome.com. The UAC would use TLS
   and cert based credentials to the first hop proxy,
   but my home proxie or UAS is going to want to see
   _my_ (as in mat@cisco.com's) credentials, not
   the UAC's. And telling me personally that I
   have to, well, use x.509 credentials is going
   to be something of a problem as I can barely
   remember a half way decent password, let alone
   ~2k characters of credentials.

   So it seems fairly obvious to me that you'd
   want to use x.509 credentials if you can, but
   the real world sez that they're not always
   going to be feasible, so you're really to have
   to deal with e2m, m2m, and e2e symmetric key
   credentials for... anything you can envision
   using asymmetric key credentials. In that case,
   TLS is providing a useful service (keeping
   outsiders of the conversation out), but doesn't
   really solve a pretty important case.

		Mike


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Michael Thomas'" <mat@cisco.com>
Cc: "'Alex Bligh'" <alex@alex.org.uk>, sip@ietf.org
Subject: RE: [Sip] RFC3261 and authenticating PROXIES to other proxies & U
	 AS - 2 possible solutions
Date: Mon, 19 Apr 2004 19:44:23 -0400
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I'm not sure you're breaking any assumption I had in mind, but...

> 
>    I can think of a pretty trivial example, I think,
>    that breaks this assumption. Suppose that I'm at
>    a public kiosk UAC and want to read my voice mail
>    back at mega.corprahome.com. The UAC would use TLS
>    and cert based credentials to the first hop proxy,
>    but my home proxie or UAS is going to want to see
>    _my_ (as in mat@cisco.com's) credentials, not
>    the UAC's. And telling me personally that I
>    have to, well, use x.509 credentials is going
>    to be something of a problem as I can barely
>    remember a half way decent password, let alone
>    ~2k characters of credentials.
> 

I would not recommend that SIP users get PKI certs to vouch for their own
identities (something, as far as I can recall, that I've never recommended).
In the passage you cited, I recommended that proxy servers get PKI certs and
use them to mutually authenticate one another on a hop-by-hop basis in SIP
trapezoids, since that was the use case under discussion in Alex's mail, and
as this would rid us of an insecurity in the proposed use of Digest for
hop-by-hop proxy server mutual authentication. This is completely orthogonal
to the end-to-end or end-to-middle authentication that you describe above.
This is just middle-to-middle.

>    So it seems fairly obvious to me that you'd
>    want to use x.509 credentials if you can, but
>    the real world sez that they're not always
>    going to be feasible, so you're really to have
>    to deal with e2m, m2m, and e2e symmetric key
>    credentials for... anything you can envision
>    using asymmetric key credentials. In that case,
>    TLS is providing a useful service (keeping
>    outsiders of the conversation out), but doesn't
>    really solve a pretty important case.
> 

TLS provides an important and feasible service even for e2m-ish cases - when
a UAC authenticates itself to a proxy server via symmetric credentials like
Digest (over TLS), the proxy server authenticates itself to the UAC with
asymmetric credentials(passed in TLS). This does not require or even suggest
that users should get their own X.509 certificates - merely that users are
able to validate certificates and signatures, and so on. This is in keeping
with the baseline RFC3261 recommendations for end-user authentication
(RFC3261 26.3.2).

> 		Mike
> 

Jon Peterson
NeuStar, Inc.

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] History-Info status - Security open issue
To: "'Mary Barnes'" <mary.barnes@nortelnetworks.com>, sip@ietf.org
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Mary,

I would support the solution you propose for progressing History-Info.

John Elwell (john.elwell@siemens.com)


 -----Original Message-----
From: Mary Barnes [mailto:mary.barnes@nortelnetworks.com] 
Sent: 16 April 2004 17:05
To: sip@ietf.org
Subject: [Sip] History-Info status - Security open issue


Hi all, 
There are still two open issues around History-Info
(http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-02.txt )
that require list discussion: privacy and security.  I'll post some details
on the privacy proposal in a separate posting so that I can more easily
track the responses.  
On the security, based upon some of the discussion in the e2m, m2e, policy
adhoc (per my recollection and the notes at:
http://softarmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc-ie
tf59.txt  ) there does not seem to be WG consensus that History-Info really
requires m2e (i.e. TLS may be sufficient).  I do agree that if there was an
m2e (and m2m) security mechanism in place for SIP at this point in time,
that the use of it for History-Info would add value and actually make
History-Info itself more useful to the end applications. 
What I would like to propose is that the draft reference the context of some
of these current works in progress in this area as informational and
possibly applicable to History-Info.  And, of course, define TLS as the
normative and mandatory security solution for History-Info. This results in
the security for History-Info initially being as robust as that of the other
headers added by Intermediaries (Via, etc.), per RFC 3261 and in the future,
could be made more robust.  In the end, it is up to the applications making
use of History-Info to choose the security solution that best meets its
requirements and if there is a m2m and m2e solution developed in the future,
applications can certainly take advantage of those.  
If there's general agreement on this approach, I believe this only requires
some minor rewording in section 3.2. The remaining work to complete the
draft would be updating and cleaning up the examples. 
[I realize that this will likely start a long thread on the more general
topic of m2e, which does need discussion, but if folks could, please,
upfront in their postings say whether they believe History-Info can and
should be progressed without that solution fully specified].
Regards, 
Mary 
mary.barnes@nortelnetworks.com 

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From: Alex Bligh <alex@alex.org.uk>
Reply-To: Alex Bligh <alex@alex.org.uk>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, sip@ietf.org
Cc: Alex Bligh <alex@alex.org.uk>
Subject: RE: [Sip] RFC3261 and authenticating PROXIES to other proxies &
 U	AS - 2 possible solutions
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Jon,

Thanks for your mail. There are several things I will have to think
further about in it, but here are some comments on bits of it:

> For the very simple SIP trapezoid case you describe below, there is of
> course a solution provided in RFC3261, specifically in 26.3.2.2: P1 and P2
> would use mutual TLS to authenticate one another. In that fashion, P2
> would have the necessary assurance that "the call came from P1", as your
> requirements below state. P1 is happy if the call came from the UAC, P2 is
> happy if it came from P1, and the UAS is happy if it came from P2 - we
> call all of this collective happiness "transitive trust". And unlike your
> solutions, it does not require that P1 and P2 exchange any keying
> information with one another.

TLS is neither a necessary nor sufficient solution, I think. Here are
some reasons.
a) In the trapezoid, it does not cover the situation where UAS wants
   to authenticate P1 (possibly against injection at P2), or where a
   third proxy P3 wants to authenticate P1. This is because TLS is
   essentially hop-by-hop. A further consequence of this is we need to
   TLS secure every possible hop between authenticator and authenticatee,
   which presupposes we know each possible route packets may take in
   advance. Certainly in the trapezoid illustration I presented, that
   is true, but I think it would be a mistake to assume hop-by-hop
   security is per-se equivalent to middle-to-middle security (or,
   in the case of UAS, end-to-middle security). This is precisely
   because we may not always be able to rely on hop-wise transitive
   trust.
b) You say P1 and P2 don't exchange any keying information; I think
   that's not entirely true. They do exchange keying, just at the TLS
   layer.
c) Using TLS requires it to be supported and switched on. Whilst this
   may well be a "good thing", there are plenty of instances in
   where it currently is unsupported, and plenty of instances in
   applications crossing administrative domains where turning it on
   is not necessarily simple. It also requires certification which
   may be a too heavyweight solution. We allow non-TLS digest
   authentication under 3261 providing the UAC is providing the
   credentials (misusing the terminology start-to-middle auth) -
   my argument is not to replace TLS, simply that the same mechanism
   could be used for middle-to-middle and middle-to-end.

> So why isn't 26.3.2.2 sufficient to solve
> all request history problems? Because sometimes transitive trust is
> insufficient. Request history also takes into account cases where the UAS,
> for example, would want to have a direct cryptographic assurance that P1
> handled the call, rather than just being content to know that P2 forwarded
> it.

I think the request-history stuff (which I have only briefly read) is
trying to solve a slightly different problem; it seems to be more about the
ability for a downstream node to be able to determine how and why a request
reached it. In the problem space I'm looking out, the downstream node (P2
or UAS) does not care how the request reached it, merely that it has passed
through P1. Further, it's eminently possible that any credential challenge
it seeks to make may be dependent on routing information available to P2
only when the packet arrives - for instance some URIs might be billable,
and some not billable. As request-history doesn't give P2 the mechanism to
challenge P1, this would seem to require P1 either giving P2 "free reign",
or having a complete understanding of P2's own authentication rules.

> Now, specifically speaking to your solutions: the first one, as this
> thread noted earlier, suffers from a transaction-matching problem. In a
> nutshell, the problem is this: if the proxy server resubmits the INVITE
> (adding a Proxy-Authorization) with a new CSeq, then the UAC will reject
> the final response from the UAS since it is out of sequence (vide RFC3261
> 22.3).

To be clear, I am advocating resubmitting with the same CSeq and a different
branch ID.

> So why shouldn't P1 just resubmit the INVITE with the old CSeq and
> a Proxy-Authorization header? Well, for the sake of argument, let's say it
> could (although I remain a little nervous that implementation snafus with
> P2 matching nonces/transactions/branches/CSeqs could arise, since RFC3261
> says that the resubmitted INVITE will arrive on a new CSeq).
>
> This approach suffers from the general problem of conflating the semantics
> of two distinct operations. Suppose that P1 resubmits the INVITE with
> credentials that are not accepted by P2, for whatever reason? How is the
> UAC going to understand that situation when the call fails?

When P1 resubmits the invite, P2 will respond with a further 407. Just as a
UAC can distinguish the "first" 407 (when it needs to add credentials, or
notes the nonce has expired) from a subsequent 407, so can P1, using the
same mechanism. On receipt of a second 407, P1 sends a 4xx upstream to the
UAC saying "call failed" - doesn't much matter which 4xx code so long as it
isn't 407 etc.; alternatively it can reroute just as it might if it
received a 404 (say) from P2. What it isn't to do is pass the 407
straight to UAC (as it will then seek credentials etc.)

> Also,
> shouldn't the UAC have its own opportunity to supply credentials to P2?
> If the user of the UAC has their own account/relationship with P2, they
> might want to rely on that rather than on allowing P1 to assert its own
> credentials - P1 is essentially interposing in the challenge.

This, it seems to me, is a policy decision for P1 and P2. P2 sees the
packet purports to come via P1 (it is unimportant for this purpose whether
it actually comes via P1), and thus use a realm identifier that P1 matches,
on the basis that P1 has asked that /it/ supply credentials to P2. If
the packet does not purport to come via P1, P2 might use a different
realm. I fail to see a useful application where P2 wishes first
to challenge P1, and then P1. I'm not saying there is no such case,
and to the extent there is, it is a limitation of this mechanism.

> Essentially, by allowing P1 to consume the 407, you are proposing a
> change to the security semantics of the Digest mechanism. If I'm
> operating a proxy server, and I send a 407, I expect that the UAC will
> handle it and resubmit. I can imagine various scenarios in which I'd be
> unhappy if that challenge were intercepted and answered by a proxy
> server.

I don't understand why. You may well be "unhappy" about it, but it
doesn't weaken security any putting it in the standard, as people
can do this /now/ (in apparent breach of the RFC). IE as P2 you already
have to cope with the situation that people might attempt to
intercept and reply to an upstream challenge. Note the intercept
will be unsuccessful unless the interceptor has valid credentials.
So if you don't like it, don't give credentials to those "in the
middle". Perhaps I am missing the point here.

> If we did want to do something like this, I imagine we'd want a
> specific mechanism (new response code, or something) that makes it
> unambiguous that P2 intends to challenge P1 - see the m2m work.

That's what I am trying to /avoid/. Take an extended trapezoid
UAC ... P1 ... P2 ... P3 ... P4 ... UAS, and let's say P4 or UAS is
performing the challenge. P3 is in the same administrative domain
as P4 & UAS. P4/UAS may not know whether it's P1, or P2 that is
going to authenticate the request. And moreover, it may not care.
All it cares is that /someone/ has authenticated the request.

If you are suggesting that there should be some additional field
in the Authorization: header, which P1 (etc.) must fill in indicating
that it is the one providing the credentials, then (provided it
isn't going to break existing challengers) that seems like a good
idea, as it will allow those challengers to discard such responses
if they want.

> Moreover, there is a security concern here. The HTTP Digest authentication
> mechanism uses the concept of 'realms' (as has previously been discussed
> in this thread). In order to prevent a malicious proxy server from
> impersonating a realm (i.e., P3 spoofs, with its own nonces and so on, a
> challenge from P2 asserting the realm of P2, in an attempt to capture or
> interrupt authentication for P2's realm), realms must be correlated with
> certificate-based security.

Perhaps I am missing something here, but doesn't this problem
exist to EXACTLY the same extent whether credentials are supplied
to P2 by P1 (as suggested), or by UAC (as per the existing standard).
IE yes, it exists, but no, the proposal doesn't make it any worse.

Moreover, they both can be solved to the same extent in the same
manner.

> In fact, the recommendations in RFC3261
> specifically state that the Digest realm should be identical to the
> subjectAltName in the certificate of the issuing proxy server - if they
> are not, there's always a potential for a security concern (26.3.2
> passim). Accordingly, the establishment of a TLS connection (and the
> resulting certificate exchange) is more or less a prerequisite for the
> secure use of HTTP Digest.

... (depending on which aspect of security you are concerned about),
yes, but equally so for conventional challenges and the challenges
I suggest.

However, unless I'm mistaken, even TLS doesn't solve everything:

   D1 :     D2A    :   D3  :    D4
      :            :       :
    --:--P1-----P2-:---P3--:--P4----P5--
   /  :            :   |   :            \
UAC   :............:   |   :             UAS
   \  :            :  /
    --:--P11---P12-:-/
      :            :
      :     D2B    :

Let's assume D1, D2A, D2B, D3, D4 are different administrative domains.
Suppose P5 challenges UAC. TLS on each hop will prevent an attack
from an arbitrary P99, but it won't (as far as I can see) prevent
(say) P2, P3 impersonating P5's realm, unless UAC has some
direct certificate exchange with P5.

And that's no greater and no lesser a problem than if P5 is challenging P1
(as I suggest), and or P5 challenging UAC.

> But of course, in your case, if certificates
> are exchanged, the use of HTTP Digest is entirely superfluous - you don't
> really care about the usernames/passwords, you just want to know which
> proxy server is forwarding the request.

No, that's not true - possibly that impression has been given by me
being less than clear. Using the above extended trapezoid, P5 wants
to know an administrative domain with whom it has a trust relationship
authorizes the routing. That might be (say) any one of P1..P2 or
their redundant siblings. So it wants to know someone from D2A
authenticated the transaction. It does indeed care about the username
(which identifies it came from D2A, as opposed to D2B). And it cares
about the password too, as that authenticates it comes from D2A as
opposed to D2B.

> You'll get that information from
> TLS anyway. So using HTTP Digest without TLS is insecure, using it with
> TLS is superfluous.

Only if it is the immediately preceding hop (unless I've missed something)

> In short, my recommendation would be just to use mutual TLS.

I am not denying that TLS is helpful - I think it's orthogonal to
the problem though, just as TLS does not replace the existing digest
authentication but is best used in parallel with it.

> Regarding your second proposed solution,

It's (the vauth header stuff) is clearly inferior to the first, and if the
request-history stuff can be authenticated, it's largely pointless. But for
the same reasons I identify above, I don't think even an authenticated
request-history solves the problems I am talking about.

Alex

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Date: Tue, 20 Apr 2004 11:35:27 +0200
From: Ali Fessi <ali.fessi@netlab.nec.de>
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Hi all,

RFC3261  provides mechanisms for authentication user agent-to-user 
agent. (Section 22.2)

The UAC is required to provide some credentials.

Although that might be not a part of the standard, but I would like to 
know how the UAs should
agree on the credentials. This assumes that the UAs "know" each other!

In some environments this might be realized with a trust third party 
that distributes the credentials.

But what is if the UAs don't have any trust relationship?!

Regards,
Ali.
--
Ali Fessi
NEC Network Laboratories





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From: Pekka Pessi <Pekka.Pessi@nokia.com>
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Subject: [Sip] Error in RFC 3261 definition of the Accept header syntax
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Hello all,

RFC 3261 defines the Accept header as follows:

20.1 Accept

   The Accept header field follows the syntax defined in [H14.1].  The
   semantics are also identical, with the exception that if no Accept
   header field is present, the server SHOULD assume a default value of
   application/sdp.

[H14.1] refers to RFC 2616 section 14.1, which plainly states that
there must be a q parameter between media-type-specific parameters
and other accept-parameters, IOW the first accept-parameter must
always be q parameter.

Now, the grammar part in RFC 3261 defines syntax for the Accept
header as follows:

Accept         =  "Accept" HCOLON
                   [ accept-range *(COMMA accept-range) ]
accept-range   =  media-range *(SEMI accept-param)
media-range    =  ( "*/*"
                  / ( m-type SLASH "*" )
                  / ( m-type SLASH m-subtype )
                  ) *( SEMI m-parameter )
accept-param   =  ("q" EQUAL qvalue) / generic-param

The error was introduced during the great grammar rewrite, I
believe. The RFC3261-style syntax following the [H14.1] would be
as follows:

Accept         =  "Accept" HCOLON
                   [ accept-range *(COMMA accept-range) ]
accept-range   =  media-range [ accept-params ]
media-range    =  ( "*/*"
                  / ( m-type SLASH "*" )
                  / ( m-type SLASH m-subtype )
                  ) *( SEMI m-parameter )
accept-params   =  SEMI "q" EQUAL qvalue *(SEMI generic-param)

--Pekka

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From exim@www1.ietf.org  Tue Apr 20 18:17:09 2004
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Alex Bligh'" <alex@alex.org.uk>, sip@ietf.org
Subject: RE: [Sip] RFC3261 and authenticating PROXIES to other proxies & U
		AS - 2 possible solutions
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Okay. At a high level, I understand that you don't think transitive trust
meets your requirements, because you envision situations in which there are
multiple intermediaries or in which there is a need for a security asseriton
between entities that are not directly connected to one another. In those
circumstances, I agree that mutual TLS is not sufficient. I do think that
mutual TLS is (by definition) sufficient for the (hopefully) common
trapezoid case in which P1 and P2 need to mutually authenticate one another.

I also understand that you feel request history is trying to solve a
different problem. I agree that request history is (possibly) broader than
what you want to do, but I do think it nicely encompasses the use cases and
requirements that you have articulated so far. While your note suggests that
Digest challenges from P2 could be customized in some fashion to the
particular circumstances of the transaction, that idea would need to be
flushed out a bit more before we can make any judgment about whether or not
request history cannot meet the underlying requirement. Whatever the
requirement is there, it could possibly serve as input to ongoing
considerations of the request history problem.

Finally, I remain unpersuaded that Digest authentication challenge
interception at proxies is really the right way to go, for two main reasons:

- I think you underestimate the complexity of pre-sharing symmetric keys for
proxy servers. Sharing symmetric keys works well as a deployment model for
the sorts of cases where usernames and passwords are successful today. For
monolithic servers like proxies, a PKI-based approach has a much cleaner
deployment model that does not suffer from any n-squared pre-sharing
problems. Proxy servers shouldn't have to have pre-shared keys in order to
be able to identify one another; their relationships are just too transient
and they may need to relate to way too many peers (unlike, say an end user's
relationship with a PSTN termination service).
- Challenge interception bends the semantics of the Digest authentication
system in ways that have foreseen undesirable consequences, and may have
unforeseen undesirable consequences. If we really need a way to do this, it
should be specific to challenging proxies, and not conflated with challenges
that are semantically designated for end users in RFC3261.

More details responses are below, for the truly dedicated.

Jon Peterson
NeuStar, Inc.

----

> TLS is neither a necessary nor sufficient solution, I think. Here are
> some reasons.
> a) In the trapezoid, it does not cover the situation where UAS wants
>    to authenticate P1 (possibly against injection at P2), or where a
>    third proxy P3 wants to authenticate P1.
<snip>

Well, as I said, I wasn't sure if you were satisfied with transitive trust,
or if you wanted to solve the full gamut of middle-to-middle problems. I was
trying to flush out your problem a bit. If the cases above are in your
requirements space, then I agree, TLS is not sufficient, and you need
something like request history / m2m. I also get the sense that you require
middle-to-end authentication as well.


> b) You say P1 and P2 don't exchange any keying information; I think
>    that's not entirely true. They do exchange keying, just at the TLS
>    layer.

I said that they do not have to pre-share their keying material with one
another. In order for Digest to work, P1 and P2 need to have shared
symmetric keys at some prior point in time. In TLS, P1 and P2 need never to
have met before in order to validate one anothers' certificates when they
are exchanged during the TLS handshake. One cannot dynamically exchange
symmetric keys in the same fashion for the purposes of Digest
authentication. In order for TLS to work, P1 and P2 only need to be in the
same PKI. There is a world of difference between the deployments for which
these alternatives are valid.

> c) Using TLS requires it to be supported and switched on. Whilst this
>    may well be a "good thing", there are plenty of instances in
>    where it currently is unsupported
<snip>

This argument is of the form "your mechanism isn't implemented or used very
widely, but my (purely conceptual and totally unimplemented) proposal
doesn't suffer from that problem." While it may be annoying to deploy a PKI,
I believe it is much more annoying to deploy and n-squared mesh of symmetric
keys to proxy servers, especially given the fact that it is very difficult
to anticipate which proxy servers are likely to need to have a relationship
with one another.

And again, just so there's no mistake, I'm talking about giving certificates
to proxy servers here, not end users. The issues with giving certificates to
end users are well known, and essentially insurmountable in our problem
space.

> 
> > So why isn't 26.3.2.2 sufficient to solve all request history problems? 
> I think the request-history stuff (which I have only briefly read) is
> trying to solve a slightly different problem; it seems to be more about
the
> ability for a downstream node to be able to determine how and why a
request
> reached it.
<snip>

But surely you agree that solving the request history problem also solves
the problem of determining whether or not a request when through a given
proxy server - it is sufficient, if not necessary. I also pointed you to the
m2m/m2e work, which is perhaps more directly applicable. The two are pretty
inextricably linked in my mind, though.

> Further, it's eminently possible that any credential challenge
> it seeks to make may be dependent on routing information available to P2
> only when the packet arrives - for instance some URIs might be billable,
> and some not billable. As request-history doesn't give P2 the mechanism to
> challenge P1, this would seem to require P1 either giving P2 "free reign",
> or having a complete understanding of P2's own authentication rules.

Well, this is a requirement I didn't understand from your previous mails,
and one I'm still not sure I understand now. In what sense might the
credential challenge be different? Isn't the purpose of this challenge to
ascertain the identity of an intermediary? If so, that identity information
will be provided by request history. The "billability" of a URI wouldn't
impact P2's challenge in any obvious way - except perhaps insofar as it
might choose not to issue a call if the transaction has no settlement
implications, or something.

Whatever it is you mean here, it sounds like you are going beyond just
knowing that a request passed through P1 on the way to P2. If there is
additional information that P1 might provide which would be instrumental in
P2's decision, that's something that should be factored into the request
history work as well. As you noted, request history can carry additional
information that helps its consumer to make decisions.

If you mean that P2 might challenge with a different realm, or something,
see my notes on realm below. I'm not really sure what else about the
challenge P2 might change in order to elicit a different response from P1,
and realms in SIP are intended to be scoped to domain, not chosen
arbitrarily.

> >
> > Suppose that P1 resubmits the INVITE with
> > credentials that are not accepted by P2, for whatever reason? How is the
> > UAC going to understand that situation when the call fails?
>
<snip> 
> On receipt of a second 407, P1 sends a 4xx upstream to the
> UAC saying "call failed" - doesn't much matter which 4xx code so long as
it
> isn't 407 etc.; alternatively it can reroute just as it might if it
> received a 404 (say) from P2. What it isn't to do is pass the 407
> straight to UAC (as it will then seek credentials etc.)
> 

This is more or less exactly my point. P1 is shielding the UAC from the
authentication system. What if P2 wants to authenticate the UAC, not P1?
What gives P1 the right to interpose? This is why, if we want to
authenticate proxies with Digest, we should use a new, unambiguous status
code - so P1 can know whether it should consume and reply to the challenge,
or whether it should forward them back. You're conflating two semantically
distinct operations.


> > If the user of the UAC has their own account/relationship with P2, they
> > might want to rely on that rather than on allowing P1 to assert its own
> > credentials - P1 is essentially interposing in the challenge.
> 
> This, it seems to me, is a policy decision for P1 and P2. 

So it's a policy decision, one that can be made by any old proxy server in
the response path, whether or not the UAC should be entitled to respond to a
407 challenge? That sets a very dangerous precedent of interventionism, from
my perspective. I can imagine several cases where I personally might have an
account on P2 (say, P2 is a proxy for a PSTN termination service), but my
local proxy server, which obstinantly insists in the face of all evidence
that it has or must have an account at P2, refuses to let me answer the
challenge. The Digest tools are intended for authentication of the UAC. If
you want a tool to authenticate middlemen, then please propose something
distinct that accomplishes that function rather than conflating this with an
existing tool which does something else entirely.

But of course, once you do, you're effectively in the m2m space, and you
should be contributing to the longstanding existing work that has been done
there.

> P2 sees the
> packet purports to come via P1 (it is unimportant for this purpose whether
> it actually comes via P1), and thus use a realm identifier that P1
matches,
> on the basis that P1 has asked that /it/ supply credentials to P2. If
> the packet does not purport to come via P1, P2 might use a different
> realm.

This is contrary to the meaning of realm as it is defined in RFC3261, which
is not as broad as the definition of realm in RFC2617.

> I fail to see a useful application where P2 wishes first
> to challenge P1, and then P1. I'm not saying there is no such case,
> and to the extent there is, it is a limitation of this mechanism.
> 
> > Essentially, by allowing P1 to consume the 407, you are proposing a
> > change to the security semantics of the Digest mechanism. If I'm
> > operating a proxy server, and I send a 407, I expect that the UAC will
> > handle it and resubmit. I can imagine various scenarios in which I'd be
> > unhappy if that challenge were intercepted and answered by a proxy
> > server.
> 
> I don't understand why. You may well be "unhappy" about it, but it
> doesn't weaken security any putting it in the standard, as people
> can do this /now/ (in apparent breach of the RFC).

"Unhappy" = my PSTN termination service might be getting revenue from a
paying customer, if the proxy at P1 were not preventing me from
authenticating my customer. Understand? I agree that this isn't a potential
security violation as such, it's an introduction of an intermediary that can
impede the operation of an existing security mechanism (as a trade-off in
order to facilitate an unrelated security operation).

SIP intermediaries today can also choose to replace the Contact header
field, the m= and c= lines of the SDP, and other aspects of messages with
the content of their choosing, in apparent breach of the RFC, as you say.
This does not mean that we should publish standards that condone any
possible course of action that a protocol entity might take to mangle SIP.

> IE as P2 you already
> have to cope with the situation that people might attempt to
> intercept and reply to an upstream challenge. Note the intercept
> will be unsuccessful unless the interceptor has valid credentials.
> So if you don't like it, don't give credentials to those "in the
> middle". Perhaps I am missing the point here.
> 

Well, in your case as described above, if the interceptor in the middle does
not have valid credentials, they send back a cryptic 4xx message to the UAC,
effectively saying "your call failed". So, if P1 does act this way, and the
UAC has valid credentials, a call fails which should not have failed.
Perhaps that is the point here.

> > If we did want to do something like this, I imagine we'd want a
> > specific mechanism (new response code, or something) that makes it
> > unambiguous that P2 intends to challenge P1 - see the m2m work.
> 
> That's what I am trying to /avoid/. Take an extended trapezoid
> UAC ... P1 ... P2 ... P3 ... P4 ... UAS, and let's say P4 or UAS is
> performing the challenge. P3 is in the same administrative domain
> as P4 & UAS. P4/UAS may not know whether it's P1, or P2 that is
> going to authenticate the request. And moreover, it may not care.
> All it cares is that /someone/ has authenticated the request.
> 

... well, that isn't how I originally understood your requirements from your
earlier mail, but okay. I think that this just broadens your problem space,
and makes it much clearer to me that you want something like RH/M2M/M2E.

Also, be careful when you say that it doesn't care who authenticated the
request. I assume you mean that it doesn't care who replied to the
challenge. But what is the purpose of replying to the challenge? To provided
credentials proving someone's identity. That is the key to this. I don't
think there's something magical about Digest as a means to prove identity.
With request history, P4 would see the identity of P1, P2 and P3, with a
cryptographic assurance, and could make its own decision whether or not any
or all of them were authorized to proxy this transaction.

> If you are suggesting that there should be some additional field
> in the Authorization: header, which P1 (etc.) must fill in indicating
> that it is the one providing the credentials, then (provided it
> isn't going to break existing challengers) that seems like a good
> idea, as it will allow those challengers to discard such responses
> if they want.

No, but I can see why that might be requirement (and the username portion of
Proxy-Authorization would no doubt make it clear who is providing the
credentials). I was thinking more that P2 should have a way to say who it
intends to challenge when it issues a 407. Today, 407 means ONLY that P2
intends to challenge the UAC. If you want to change that, then we need some
way of qualifying the challenge to indicate that someone other than the UAC
should answer it.

> 
> > Moreover, there is a security concern here. The HTTP Digest 
> authentication
> > mechanism uses the concept of 'realms' (as has previously 
> been discussed
> > in this thread). In order to prevent a malicious proxy server from
> > impersonating a realm (i.e., P3 spoofs, with its own nonces 
> and so on, a
> > challenge from P2 asserting the realm of P2, in an attempt 
> to capture or
> > interrupt authentication for P2's realm), realms must be 
> correlated with
> > certificate-based security.
> 
> Perhaps I am missing something here, but doesn't this problem
> exist to EXACTLY the same extent whether credentials are supplied
> to P2 by P1 (as suggested), or by UAC (as per the existing standard).
> IE yes, it exists, but no, the proposal doesn't make it any worse.
> 
> Moreover, they both can be solved to the same extent in the same
> manner.
> 

Of course. Your proposal does not make this problem any worse. It was raised
to show that your solution is unnecessary in transitive trapezoid cases. If
your solution space goes beyond transitive cases, though, then this will not
apply. You clearly seem to want a direct assurance, rather than a transitive
assurance, so I agree, mutual TLS is not sufficent for that.

<big snip of discussion of whether or not your case is transitive or direct
assurance>
> 
> I am not denying that TLS is helpful - I think it's orthogonal to
> the problem though, just as TLS does not replace the existing digest
> authentication but is best used in parallel with it.
> 

In trapezoid cases where your interest is in P2 determining that a request
was received via P1, it is sufficient. In that sense, it's not orthogonal to
the problem - the issue is that it is only valid for that one case (though
it is a case that seems to be a common one). Again, you clearly want to
cover other spaces.

> > Regarding your second proposed solution,
> 
> It's (the vauth header stuff) is clearly inferior to the first, and if the
> request-history stuff can be authenticated, it's largely pointless. But
for
> the same reasons I identify above, I don't think even an authenticated
> request-history solves the problems I am talking about.
> 

Above, I did not see any reason that request-history does not solve your
problem. You argued, perhaps, that request-history is broader than your
problem, but that doesn't mean that it doesn't solve it. If you have some
additional requirements for request history that have not yet been captured
(whatever flavor you think you might get from a dynamic challenge from P2),
those should be submitted for consideration to the request history work.

> Alex
> 

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From: Alex Bligh <alex@alex.org.uk>
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To: "Peterson, Jon" <jon.peterson@neustar.biz>, sip@ietf.org
Cc: Alex Bligh <alex@alex.org.uk>
Subject: RE: [Sip] RFC3261 and authenticating PROXIES to other proxies &
 U		AS - 2 possible solutions
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Jon,

> I also understand that you feel request history is trying to solve a
> different problem. I agree that request history is (possibly) broader than
> what you want to do, but I do think it nicely encompasses the use cases
> and requirements that you have articulated so far.

The requirements possibly, but it doesn't actually provide a solution (see
draft-barnes-sipping-sec-inserted-info-01.txt section 4. for why :-) ). Of
course this is not to say it couldn't provide a solution, but it is a bit
difficult to compare a concrete proposal to what in essence is a framework
stating a problem, but not yet proposing a solution.

Note that I am in no way trying to denigrate either the request history
work or the sec-inserted-info work; all I am saying is that a simple
change will provide a pragmatic solution to a particular problem space.
Not in either as general terms or as secure terms as you are proposing,
but in a manner which works /currently/ with all devices producing
a challenge (as far as I can see).

> While your note
> suggests that Digest challenges from P2 could be customized in some
> fashion to the particular circumstances of the transaction, that idea
> would need to be flushed out a bit more before we can make any judgment
> about whether or not request history cannot meet the underlying
> requirement.

See below.


> - I think you underestimate the complexity of pre-sharing symmetric keys
> for proxy servers. Sharing symmetric keys works well as a deployment
> model for the sorts of cases where usernames and passwords are successful
> today. For monolithic servers like proxies, a PKI-based approach has a
> much cleaner deployment model that does not suffer from any n-squared
> pre-sharing problems. Proxy servers shouldn't have to have pre-shared
> keys in order to be able to identify one another; their relationships are
> just too transient and they may need to relate to way too many peers
> (unlike, say an end user's relationship with a PSTN termination service).

I am not sure why the problem is n-squared. In all the examples of problems
in the space this solution is proposed to solve, there is a 1:few problem,
and more relevantly the authentication relationship is a "symptom" of some
greater trust relationship that has already been set up. For instance, in
the "my proxy diverts incoming SIP calls to me to my cellphone when my SIP
UAS is unavailble" environment, there is already some form of
(administrative) trust relationship between the SIP proxy and the gateway
used to perform the diversion.

You are entirely correct that digest authentication does not and could
not solve the general problem of how a given proxy P(n) authenticates
that any and all intermediate notes P(1)..P(n-1) are who they say they
are. However, I am not sure of what problem that is a solution to.

> - Challenge interception bends the semantics of the Digest authentication
> system in ways that have foreseen undesirable consequences, and may have
> unforeseen undesirable consequences.

I am still not sure what the /foreseen/ undesirable consequences are
(modulo those that you mention below which I have sought to address).
Any protocol change might have unforeseen undesirable consequences,
but that is not per-se an excuse not to ever change any protocols. The
key it seems to me is minimizing the risk of ill-effect, and getting
as many bright people on this list and elsewhere to look through and
say "well I can't think of any".

> If we really need a way to do this,
> it should be specific to challenging proxies, and not conflated with
> challenges that are semantically designated for end users in RFC3261.

I respectfully disagree with you there. If you can use a general mechanism
safely, rather than 2 specific mechanisms, the former seems better. If
there is an inherent danger in using a general mechanism, then that
should be avoided, but in the absence of that danger, I would suggest
as much commonality as possible is a good thing in terms of protocol
simplification.
> In order for TLS to work, P1 and P2 only need to be in the
> same PKI.

Sure, but then you have added the additional necessity of full PKI
certificates in order to provide authentication by a proxy. Whilst
I can see why this may be useful in some circumstances, in a real
world deployment, that sounds pretty heavyweight if the security
of a simple digest challenge would be sufficient.

>> c) Using TLS requires it to be supported and switched on. Whilst this
>>    may well be a "good thing", there are plenty of instances in
>>    where it currently is unsupported
> <snip>
>
> This argument is of the form "your mechanism isn't implemented or used
> very widely, but my (purely conceptual and totally unimplemented) proposal
> doesn't suffer from that problem."

:-)

Whilst there is some truth in that, I think you've missed the point.
My (purely conceptual) proposal is already currently implemented on
every device (UAS and Proxy) which can provide a challenge. It's
only unimplemented on those that need to respond to the challenge.
Having looked at the code for ser (and, admittedly, nothing else),
it doesn't seem like too great a change to me - I would suggest
rather less of a change than to add (say) TLS and appropriate PKI.

Besides which, your proposed solution that /does/ fully address the
problem, IE the request history stuff, if the security aspect is maximally
documented at in draft-barnes-sipping-sec-inserted-info-01.txt, is not even
*DESCRIBED* in full yet , thus cannot be implemented or used at all.
Again, I am not in any way denigrating the work, I am just saying from
a pragmatic point of view what I am suggesting is a minimal change
fix.

> While it may be annoying to deploy a
> PKI, I believe it is much more annoying to deploy and n-squared mesh of
> symmetric keys to proxy servers, especially given the fact that it is
> very difficult to anticipate which proxy servers are likely to need to
> have a relationship with one another.

There is no argument that if the problem space demands an n-squared
mesh, PKI is better than any digest system (and, for authentication
of UACs too). However, I don't believe this (requirement for
n to m authentication) is the case.

> But surely you agree that solving the request history problem also solves
> the problem of determining whether or not a request when through a given
> proxy server - it is sufficient, if not necessary. I also pointed you to
> the m2m/m2e work, which is perhaps more directly applicable. The two are
> pretty inextricably linked in my mind, though.

Well it might well solve that problem. But it's difficult to see
whether it actually does or will solve that problem as the security
stuff is at quite an early stage.

>> Further, it's eminently possible that any credential challenge
>> it seeks to make may be dependent on routing information available to P2
>> only when the packet arrives - for instance some URIs might be billable,
>> and some not billable. As request-history doesn't give P2 the mechanism
>> to challenge P1, this would seem to require P1 either giving P2 "free
>> reign", or having a complete understanding of P2's own authentication
>> rules.
>
> Well, this is a requirement I didn't understand from your previous mails,
> and one I'm still not sure I understand now. In what sense might the
> credential challenge be different? Isn't the purpose of this challenge to
> ascertain the identity of an intermediary?

To ascertain the that an intermediary has appropriate credentials.
IE I don't necessarily care which server from an authorized intermediate
administrative domain inserted them, I just care they have a username
and password tuple.

If I can tell only that it is a specific intermediate server, I then
have to map that back to an appropriate administrative domain etc.;
not necessarily a problem, but it does mean if administrative domain
X has 100 different servers, all with different certificates, then
proxy server Y outside there has to have each of the 100 different
certificates (or have some other way of mapping each of the ID's
to administrative domain X).

> If so, that identity
> information will be provided by request history. The "billability" of a
> URI wouldn't impact P2's challenge in any obvious way - except perhaps
> insofar as it might choose not to issue a call if the transaction has no
> settlement implications, or something.

What I was suggesting is the information P1 might provide (or chose not
to provide) might depend on information P2 would pass it in the challenge.
As relatively trivial examples, it could pass the cost of the call,
the QoS it could provide, the RTT etc. It needn't be passed in the
realm necessarily. P1 might decide not to authenticate the request
based on this information.

>> On receipt of a second 407, P1 sends a 4xx upstream to the
>> UAC saying "call failed" - doesn't much matter which 4xx code so long as
> it
>> isn't 407 etc.; alternatively it can reroute just as it might if it
>> received a 404 (say) from P2. What it isn't to do is pass the 407
>> straight to UAC (as it will then seek credentials etc.)
>>
>
> This is more or less exactly my point. P1 is shielding the UAC from the
> authentication system. What if P2 wants to authenticate the UAC, not P1?
> What gives P1 the right to interpose? This is why, if we want to
> authenticate proxies with Digest, we should use a new, unambiguous status
> code - so P1 can know whether it should consume and reply to the
> challenge, or whether it should forward them back. You're conflating two
> semantically distinct operations.

The advantage of using the same status code is that both proxies and
UAC's can authenticate to the same proxy. Very useful in the case of
(say) a media gateway service where some of its subscribers are UAs,
some are (effectively) B2BUAs or media-gateway-esque devices (asterisk
for example), and some are other proxies. As the proxy does not at this
stage know who the call came from, how would it know to respond with
"Proxy Authentication Required" or your new code?

However, saying that I do think you make a reasonable point that there
should be some mechanism to forbid an intervening proxy from intercepting
(perhaps a perjorative word) the 407. I am unsure whether it would
be best that an originating UAC said "No, *I* want to authenticate
this call", or whether the proxy should be saying "I want this
to go right back to the UAC".

>> > If the user of the UAC has their own account/relationship with P2, they
>> > might want to rely on that rather than on allowing P1 to assert its own
>> > credentials - P1 is essentially interposing in the challenge.
>>
>> This, it seems to me, is a policy decision for P1 and P2.
>
> So it's a policy decision, one that can be made by any old proxy server in
> the response path, whether or not the UAC should be entitled to respond
> to a 407 challenge?

The UAC is always entitled to respond. The question is whether an
intervening proxy is entitled to "intercept" (as you call it). And my
answer is that this MUST NOT be done UNLESS there is a preexisting
trust relationship between the proxy issuing the challenge and the
proxy replying to the challenge. That is not "any old proxy server".
That is a proxy server which the administrators of both systems
have agreed should respond (as it has credentials).


>> P2 sees the
>> packet purports to come via P1 (it is unimportant for this purpose
>> whether it actually comes via P1), and thus use a realm identifier that
>> P1
> matches,
>> on the basis that P1 has asked that /it/ supply credentials to P2. If
>> the packet does not purport to come via P1, P2 might use a different
>> realm.
>
> This is contrary to the meaning of realm as it is defined in RFC3261,
> which is not as broad as the definition of realm in RFC2617.

Are you saying that under RFC3261 I can't configure a single machine
to carry two security certificates (i.e. act as sip.mynet.com and
sip.mywhitelabel.com)?

>> I don't understand why. You may well be "unhappy" about it, but it
>> doesn't weaken security any putting it in the standard, as people
>> can do this /now/ (in apparent breach of the RFC).
>
> "Unhappy" = my PSTN termination service might be getting revenue from a
> paying customer, if the proxy at P1 were not preventing me from
> authenticating my customer. Understand?

Nope I don't understand. P1 is only "preventing" you from authenticating
your customer as you (P2) have previously come to an agreement with
P1 that P1 is your customer (which does not exclude UAC being your
customer too), and P1 is thus attempting to give you customer traffic
as per a previously established customer relationship.

P1 must not arbitrarily respond to realms "on the off chance" it can
authenticated the transaction.

> Well, in your case as described above, if the interceptor in the middle
> does not have valid credentials, they send back a cryptic 4xx message to
> the UAC, effectively saying "your call failed". So, if P1 does act this
> way, and the UAC has valid credentials, a call fails which should not
> have failed. Perhaps that is the point here.

I am not sure the above call should NOT have failed. Let's take the case
where UAC makes a SIP call via P1 and P2 to a switched off UASX, and P2
wants to divert the call to a cellphone via P3 and a media-gw.

The idea here is that P3 or media-gw (UAS) authenticates the call came via
P2, as P2 (providing service to UAS) is going to be billed for it.

UAC thinks the call is going to cost whatever a SIP call normally costs. If
P2's authentication works, then indeed the call should proceed.

However, let's suppose P2's authentication fails (perhaps they've forgotten
to pay their bill), and by some chance, UAC also has credentials with P3
(perhaps P3 is actually the same administrative domain as P1, i.e. UAC's
provider). Then automagically UAC will supply its own credentials to P3,
the call will get through, but UAC will be billed for the privilege. This
was exactly what UASX sought to avoid.

The message back need only be as cryptic or descriptive as P2 wants it to
be. If the call went to a media-gateway that hadn't paid its POTS bill, it
will get a far more cryptic message back.

> ... well, that isn't how I originally understood your requirements from
> your earlier mail, but okay. I think that this just broadens your problem
> space, and makes it much clearer to me that you want something like
> RH/M2M/M2E.
>
> Also, be careful when you say that it doesn't care who authenticated the
> request. I assume you mean that it doesn't care who replied to the
> challenge.

I mean I care only that valid credentials identifying are presented
(whether they are used to identify individual servers, or administrative
domains is not necessarily relevant). In many applications it is
unnecessary to identify individual nodes.

> But what is the purpose of replying to the challenge? To
> provided credentials proving someone's identity. That is the key to this.
> I don't think there's something magical about Digest as a means to prove
> identity. With request history, P4 would see the identity of P1, P2 and
> P3, with a cryptographic assurance, and could make its own decision
> whether or not any or all of them were authorized to proxy this
> transaction.

This would indeed be true, if the specification and the software
to implement it existed. What I am proposing (a) has a specification,
(b) has one end of the software already written.

>> If you are suggesting that there should be some additional field
>> in the Authorization: header, which P1 (etc.) must fill in indicating
>> that it is the one providing the credentials, then (provided it
>> isn't going to break existing challengers) that seems like a good
>> idea, as it will allow those challengers to discard such responses
>> if they want.
>
> No, but I can see why that might be requirement (and the username portion
> of Proxy-Authorization would no doubt make it clear who is providing the
> credentials). I was thinking more that P2 should have a way to say who it
> intends to challenge when it issues a 407. Today, 407 means ONLY that P2
> intends to challenge the UAC. If you want to change that, then we need
> some way of qualifying the challenge to indicate that someone other than
> the UAC should answer it.

Perhaps "noone other than UAC should answer it".

> Above, I did not see any reason that request-history does not solve your
> problem. You argued, perhaps, that request-history is broader than your
> problem, but that doesn't mean that it doesn't solve it. If you have some
> additional requirements for request history that have not yet been
> captured (whatever flavor you think you might get from a dynamic
> challenge from P2), those should be submitted for consideration to the
> request history work.

This argument sounds a bit like "we cannot solve a small isolated problem
by generalizing an existing mechanism, if we have the opportunity of
solving at a rather later date a much larger more complicated problem which
a rather heavier-weight solution that will probably solve the small problem
too". That's one of the things that broke OSI, X.400 etc.; not doing that
made IP et al. work.

I am not suggesting this replace request-history etc. - there is
no reason the two cannot live together in harmony. I will take
your word for it than when request-history & appropriate security
stuff is written you can use it to satisfactorily authenticate a
request has passed through a proxy, but it seems a bit like a
sledgehammer to crack a nut.

Alex

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User-Agent: Microsoft-Entourage/10.1.4.030702.0
Date: Wed, 21 Apr 2004 15:03:38 -0700
Subject: Re: [Sip] History-Info Privacy Issue
From: Cullen Jennings <fluffy@cisco.com>
To: Mary Barnes <mary.barnes@nortelnetworks.com>, <sip@ietf.org>
Message-ID: <BCAC3DCA.3A7DB%fluffy@cisco.com>
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I don=B9t care a huge amount but I think I slightly favor 2.


On 4/16/04 9:05 AM, "Mary Barnes" <mary.barnes@nortelnetworks.com> wrote:

> Per my posting on the status of History-Info, Privacy is one of the remai=
ning
> open issues.  The proposal currently in the draft is quite simple:
>=20
> If the Privacy header has a priv-value of "Session" or "Header" privacy, =
the
> History-Info  SHOULD not be included in the Request (or subsequent respon=
ses).
>=20
> This is effectively applying the fundamental recommendation in RFC3323 th=
at
> any optional information that can divulge information about the user(s) s=
hould
> not be included in the requests.
>=20
> A proposal was made that there are scenarios (e.g. where local policy lim=
its
> the applicability of History-Info within a specific domain) where you wou=
ld
> want to capture the History-Info but not send it beyond certain intermedi=
aries
> or domains.  To support this, it was proposed that we could extend the Pr=
ivacy
> header defined in RFC 3323 with tags for History-Info something like the
> following:
>=20
> priv-value =3D "history" / "history-index"
>=20
> The "history" priv-value would mean that all the History-Info entries sho=
uld
> be removed.  The "history-index" would be used to indicate that the speci=
fic
> entry is to be removed and would need to have the Privacy header escaped =
in
> the URI so that it can be associated with a specific History-Info entry (=
or we
> need to add another field to History-Info for this).
>=20
> So, I think we have two options here:
> 1) Extend History-Info to have an optional privacy field along the lines =
of
> what's detailed above.
>=20
> 2) Defer the definition of the privacy field to a separate draft, since i=
t
> appears to be primarily limited to scenarios with certain trust models.
>=20
> So, if folks could just take a bit of time and consider their preference,=
 I
> would like to get the draft updated soon and closed off prior to the Inte=
rim
> meeting.
>=20
> Regards,=20
> Mary H. Barnes=20
> mary.barnes@nortelnetworks.com
>=20



--B_3165404618_122344
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<HTML>
<HEAD>
<TITLE>Re: [Sip] History-Info Privacy Issue</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
I don&#8217;t care a huge amount but I think I slightly favor 2. <BR>
<BR>
<BR>
On 4/16/04 9:05 AM, &quot;Mary Barnes&quot; &lt;mary.barnes@nortelnetworks.=
com&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0p=
x'>Per my posting on the status of History-Info, Privacy is one of the remai=
ning open issues. &nbsp;The proposal currently in the draft is quite simple:=
 <BR>
<BR>
If the Privacy header has a priv-value of &quot;Session&quot; or &quot;Head=
er&quot; privacy, the History-Info &nbsp;SHOULD not be included in the Reque=
st (or subsequent responses). &nbsp;<BR>
<BR>
This is effectively applying the fundamental recommendation in RFC3323 that=
 any optional information that can divulge information about the user(s) sho=
uld not be included in the requests. &nbsp;<BR>
<BR>
A proposal was made that there are scenarios (e.g. where local policy limit=
s the applicability of History-Info within a specific domain) where you woul=
d want to capture the History-Info but not send it beyond certain intermedia=
ries or domains. &nbsp;To support this, it was proposed that we could extend=
 the Privacy header defined in RFC 3323 with tags for History-Info something=
 like the following:<BR>
<BR>
priv-value =3D &quot;history&quot; / &quot;history-index&quot; &nbsp;<BR>
<BR>
The &quot;history&quot; priv-value would mean that all the History-Info ent=
ries should be removed. &nbsp;The &quot;history-index&quot; would be used to=
 indicate that the specific entry is to be removed and would need to have th=
e Privacy header escaped in the URI so that it can be associated with a spec=
ific History-Info entry (or we need to add another field to History-Info for=
 this). &nbsp;&nbsp;&nbsp;<BR>
<BR>
So, I think we have two options here: <BR>
1) Extend History-Info to have an optional privacy field along the lines of=
 what's detailed above. &nbsp;<BR>
<BR>
2) Defer the definition of the privacy field to a separate draft, since it =
appears to be primarily limited to scenarios with certain trust models. &nbs=
p;<BR>
<BR>
So, if folks could just take a bit of time and consider their preference, I=
 would like to get the draft updated soon and closed off prior to the Interi=
m meeting.<BR>
<BR>
Regards, <BR>
Mary H. Barnes <BR>
mary.barnes@nortelnetworks.com <BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT>
</BODY>
</HTML>


--B_3165404618_122344--


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Date: Wed, 21 Apr 2004 14:59:28 -0700
Subject: Re: [Sip] History-Info status - Security open issue
From: Cullen Jennings <fluffy@cisco.com>
To: Mary Barnes <mary.barnes@nortelnetworks.com>, <sip@ietf.org>
Message-ID: <BCAC3CD0.3A7C4%fluffy@cisco.com>
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Yes. Sounds reasonable to me.

On 4/16/04 9:04 AM, "Mary Barnes" <mary.barnes@nortelnetworks.com> wrote:

> Hi all, 
> 
> There are still two open issues around History-Info
> (http://www.ietf.org/internet-drafts/draft-ietf-sip-history-info-02.txt ) that
> require list discussion: privacy and security.  I'll post some details on the
> privacy proposal in a separate posting so that I can more easily track the
> responses.  
> 
> On the security, based upon some of the discussion in the e2m, m2e, policy
> adhoc (per my recollection and the notes at:
> http://softarmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc-ietf
> 59.txt  ) there does not seem to be WG consensus that History-Info really
> requires m2e (i.e. TLS may be sufficient).  I do agree that if there was an
> m2e (and m2m) security mechanism in place for SIP at this point in time, that
> the use of it for History-Info would add value and actually make History-Info
> itself more useful to the end applications.
> 
> What I would like to propose is that the draft reference the context of some
> of these current works in progress in this area as informational and possibly
> applicable to History-Info.  And, of course, define TLS as the normative and
> mandatory security solution for History-Info. This results in the security for
> History-Info initially being as robust as that of the other headers added by
> Intermediaries (Via, etc.), per RFC 3261 and in the future, could be made more
> robust.  In the end, it is up to the applications making use of History-Info
> to choose the security solution that best meets its requirements and if there
> is a m2m and m2e solution developed in the future, applications can certainly
> take advantage of those.
> 
> If there's general agreement on this approach, I believe this only requires
> some minor rewording in section 3.2. The remaining work to complete the draft
> would be updating and cleaning up the examples.
> 
> [I realize that this will likely start a long thread on the more general topic
> of m2e, which does need discussion, but if folks could, please, upfront in
> their postings say whether they believe History-Info can and should be
> progressed without that solution fully specified].
> 
> Regards, 
> Mary 
> mary.barnes@nortelnetworks.com
> 
> 



--B_3165404369_109628
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<HTML>
<HEAD>
<TITLE>Re: [Sip] History-Info status - Security open issue</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
Yes. Sounds reasonable to me. <BR>
<BR>
On 4/16/04 9:04 AM, &quot;Mary Barnes&quot; &lt;mary.barnes@nortelnetworks.=
com&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0p=
x'>Hi all, <BR>
<BR>
There are still two open issues around History-Info (<a href=3D"http://www.ie=
tf.org/internet-drafts/draft-ietf-sip-history-info-02.txt">http://www.ietf.o=
rg/internet-drafts/draft-ietf-sip-history-info-02.txt</a> ) that require lis=
t discussion: privacy and security. &nbsp;I'll post some details on the priv=
acy proposal in a separate posting so that I can more easily track the respo=
nses. &nbsp;<BR>
<BR>
On the security, based upon some of the discussion in the e2m, m2e, policy =
adhoc (per my recollection and the notes at: <a href=3D"http://softarmor.com/s=
ipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc-ietf59.txt">http://softa=
rmor.com/sipping/meets/ietf59/notes/notes-cyrus-sipping-adhoc-ietf59.txt</a>=
 &nbsp;) there does not seem to be WG consensus that History-Info really req=
uires m2e (i.e. TLS may be sufficient). &nbsp;I do agree that if there was a=
n m2e (and m2m) security mechanism in place for SIP at this point in time, t=
hat the use of it for History-Info would add value and actually make History=
-Info itself more useful to the end applications. <BR>
<BR>
What I would like to propose is that the draft reference the context of som=
e of these current works in progress in this area as informational and possi=
bly applicable to History-Info. &nbsp;And, of course, define TLS as the norm=
ative and mandatory security solution for History-Info. This results in the =
security for History-Info initially being as robust as that of the other hea=
ders added by Intermediaries (Via, etc.), per RFC 3261 and in the future, co=
uld be made more robust. &nbsp;In the end, it is up to the applications maki=
ng use of History-Info to choose the security solution that best meets its r=
equirements and if there is a m2m and m2e solution developed in the future, =
applications can certainly take advantage of those. &nbsp;<BR>
<BR>
If there's general agreement on this approach, I believe this only requires=
 some minor rewording in section 3.2. The remaining work to complete the dra=
ft would be updating and cleaning up the examples. <BR>
<BR>
[I realize that this will likely start a long thread on the more general to=
pic of m2e, which does need discussion, but if folks could, please, upfront =
in their postings say whether they believe History-Info can and should be pr=
ogressed without that solution fully specified].<BR>
<BR>
Regards, <BR>
Mary <BR>
mary.barnes@nortelnetworks.com <BR>
<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
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From exim@www1.ietf.org  Thu Apr 22 10:02:42 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: fluffy@cisco.com, mary.barnes@nortelnetworks.com, sip@ietf.org
Subject: AW: [Sip] History-Info Privacy Issue
Date: Thu, 22 Apr 2004 15:19:52 +0200
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I favour 1, because I think this should be kept together.
=20
Best Regards
=20
Roland

-----Urspr=FCngliche Nachricht-----
Von: sip-admin@ietf.org [mailto:sip-admin@ietf.org]Im Auftrag von =
Cullen Jennings
Gesendet: Donnerstag, 22. April 2004 00:04
An: Mary Barnes; sip@ietf.org
Betreff: Re: [Sip] History-Info Privacy Issue



I don't care a huge amount but I think I slightly favor 2.=20


On 4/16/04 9:05 AM, "Mary Barnes" <mary.barnes@nortelnetworks.com> =
wrote:



Per my posting on the status of History-Info, Privacy is one of the =
remaining open issues.  The proposal currently in the draft is quite =
simple:=20

If the Privacy header has a priv-value of "Session" or "Header" =
privacy, the History-Info  SHOULD not be included in the Request (or =
subsequent responses). =20

This is effectively applying the fundamental recommendation in RFC3323 =
that any optional information that can divulge information about the =
user(s) should not be included in the requests. =20

A proposal was made that there are scenarios (e.g. where local policy =
limits the applicability of History-Info within a specific domain) =
where you would want to capture the History-Info but not send it beyond =
certain intermediaries or domains.  To support this, it was proposed =
that we could extend the Privacy header defined in RFC 3323 with tags =
for History-Info something like the following:

priv-value =3D "history" / "history-index" =20

The "history" priv-value would mean that all the History-Info entries =
should be removed.  The "history-index" would be used to indicate that =
the specific entry is to be removed and would need to have the Privacy =
header escaped in the URI so that it can be associated with a specific =
History-Info entry (or we need to add another field to History-Info for =
this).   =20

So, I think we have two options here:=20
1) Extend History-Info to have an optional privacy field along the =
lines of what's detailed above. =20

2) Defer the definition of the privacy field to a separate draft, since =
it appears to be primarily limited to scenarios with certain trust =
models. =20

So, if folks could just take a bit of time and consider their =
preference, I would like to get the draft updated soon and closed off =
prior to the Interim meeting.

Regards,=20
Mary H. Barnes=20
mary.barnes@nortelnetworks.com=20







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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Re: [Sip] History-Info Privacy Issue</TITLE>

<META content=3D"MSHTML 5.50.4933.1800" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
favour 1, because I think this should be kept =
together.</FONT></SPAN></DIV>
<DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff size=3D2>Best=20
Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>Roland</FONT></SPAN></DIV>
<BLOCKQUOTE>
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> =
sip-admin@ietf.org=20
  [mailto:sip-admin@ietf.org]<B>Im Auftrag von </B>Cullen=20
  Jennings<BR><B>Gesendet:</B> Donnerstag, 22. April 2004 =
00:04<BR><B>An:</B>=20
  Mary Barnes; sip@ietf.org<BR><B>Betreff:</B> Re: [Sip] History-Info =
Privacy=20
  Issue<BR><BR></FONT></DIV><FONT face=3DVerdana><SPAN=20
  style=3D"FONT-SIZE: 14px"><BR>I don't care a huge amount but I think =
I slightly=20
  favor 2. <BR><BR><BR>On 4/16/04 9:05 AM, "Mary Barnes"=20
  &lt;mary.barnes@nortelnetworks.com&gt; wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3DVerdana><SPAN style=3D"FONT-SIZE: 14px">Per =
my posting=20
    on the status of History-Info, Privacy is one of the remaining open =
issues.=20
    &nbsp;The proposal currently in the draft is quite simple: =
<BR><BR>If the=20
    Privacy header has a priv-value of "Session" or "Header" privacy, =
the=20
    History-Info &nbsp;SHOULD not be included in the Request (or =
subsequent=20
    responses). &nbsp;<BR><BR>This is effectively applying the =
fundamental=20
    recommendation in RFC3323 that any optional information that can =
divulge=20
    information about the user(s) should not be included in the =
requests.=20
    &nbsp;<BR><BR>A proposal was made that there are scenarios (e.g. =
where local=20
    policy limits the applicability of History-Info within a specific =
domain)=20
    where you would want to capture the History-Info but not send it =
beyond=20
    certain intermediaries or domains. &nbsp;To support this, it was =
proposed=20
    that we could extend the Privacy header defined in RFC 3323 with =
tags for=20
    History-Info something like the following:<BR><BR>priv-value =3D =
"history" /=20
    "history-index" &nbsp;<BR><BR>The "history" priv-value would mean =
that all=20
    the History-Info entries should be removed. &nbsp;The =
"history-index" would=20
    be used to indicate that the specific entry is to be removed and =
would need=20
    to have the Privacy header escaped in the URI so that it can be =
associated=20
    with a specific History-Info entry (or we need to add another field =
to=20
    History-Info for this). &nbsp;&nbsp;&nbsp;<BR><BR>So, I think we =
have two=20
    options here: <BR>1) Extend History-Info to have an optional =
privacy field=20
    along the lines of what's detailed above. &nbsp;<BR><BR>2) Defer =
the=20
    definition of the privacy field to a separate draft, since it =
appears to be=20
    primarily limited to scenarios with certain trust models. =
&nbsp;<BR><BR>So,=20
    if folks could just take a bit of time and consider their =
preference, I=20
    would like to get the draft updated soon and closed off prior to =
the Interim=20
    meeting.<BR><BR>Regards, <BR>Mary H. Barnes=20
    <BR>mary.barnes@nortelnetworks.com =
<BR><BR></SPAN></FONT></BLOCKQUOTE><FONT=20
  face=3DVerdana><SPAN=20
style=3D"FONT-SIZE: 14px"><BR></BLOCKQUOTE></SPAN></FONT></BODY></HTML>

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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From exim@www1.ietf.org  Thu Apr 22 17:43:05 2004
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Date: Thu, 22 Apr 2004 14:51:23 -0500
To: Hisham Khartabil <hisham.khartabil@nokia.com>,
        Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>
From: "James M. Polk" <jmpolk@cisco.com>
Cc: sipping@ietf.org, sip@ietf.org
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Subject: [Sip] Re: SIMPLE/SIP/SIPPING/XCON Working Groups Joint Interim
 Meeting Announcement
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What happened to time for Emergency and E2M Security? In Seoul, these were 
mentioned as two of the prevailing topic areas for the interim (along with 
Session Policy and Exploders).

At 11:31 AM 4/22/2004 -0400, Hisham Khartabil wrote:
>An joint interim meeting for the SIMPLE, SIP, SIPPING and XCON working groups
>will be help from Monday 24th May until Wednesday 26th May in Boston, 
>Massachusetts.
>
>Location:
>Renaissance Boston Hotel Bedford
>44 Middlesex Turnpike Bedford, MA 01730 USA
>Phone: 1 781-275-5500 Fax: 1 781-275-3042
>
>http://marriott.com/property/propertyPage.mi?marshaCode=BOSSB
>
>Agenda:
>
>MONDAY - SIMPLE
>
>9:00-12:00 - MSRP/MSRP Relays
>12:00-13:00 - Lunch
>13:00-16:00 - XCAP
>16:00-16:30 - Filtering
>16:30-17:00 - Partial Publish
>17:00-17:30 - Advanced Messaging
>
>TUESDAY - SIP/SIPPING
>
>9:00-10:30 - KPML
>10:30-11:00 - GRUU
>11:00-12:00 - Session Policies
>12:00-13:00 - Lunch
>13:00-14:30 - Session Policies Cont.
>14:30-15:30 - Exploders
>time permitting:
>15:30-16:15 - Connected Party
>16:15-17:00 - Application Interaction
>17:00-17:30 - Dialog Package
>17:30-18:00 - Connection Re-use
>
>WEDNESDAY - XCON
>
>9:00-11:00 - Floor Control
>11:00-12:00 - CPCP
>12:00-13:00 - Lunch
>13:00-14:00 - CPCP Cont.
>14:00-16:00 - MPCP
>
>The chairs would like to thank Nokia for sponsoring the event.
>
>
>
>_______________________________________________
>IETF-Announce mailing list
>IETF-Announce@ietf.org
>https://www1.ietf.org/mailman/listinfo/ietf-announce


cheers,
James

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


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Hello,
=20
Can someone please point me to the known issues with SIP security?=20
=20
Also are there any initiatives in progress for using Liberty alliance
like identity/trust infrastructure in conjunction with SIP protocols to
facilitate spontaneous trusted real time communication between a A party
and a B party, who are not registered with one SIP proxy?=20
=20
Thanks,
Anir

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initiatives in progress for using Liberty alliance like identity/trust
infrastructure in conjunction with SIP protocols to facilitate =
spontaneous trusted
real time communication between a <span class=3DSpellE>A</span> party =
and a B
party, who are not registered with one SIP proxy? =
<o:p></o:p></span></font></p>

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Subject: [Sip] How should i apprehend this sentence?
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in the section 8.2.2.1 of RFC3261,there is a sentence "Other potential=20
sources of received Request-URIs include the Contact header fields of=20
requests and responses sent by the UA that establish or refresh dialogs."
How should i apprehend it?
thx

_________________________________________________________________
=C3=E2=B7=D1=CF=C2=D4=D8 MSN Explorer:   http://explorer.msn.com/lccn =20


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From: Ali Fessi <ali.fessi@netlab.nec.de>
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To: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
Cc: sip@ietf.org
Subject: Re: [Sip] [Query] SIP security and trusted spontaneous interactions
 between end points
References: <1FD836F38652D211A63E0008C7244343125171F1@ntmsg0008.corpmail.telstra.com.au>
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Hi Anir,

to be clear, I'm not an expert! But I have been also working on SIP 
security lately. (I have been mostly working on different solutions for 
SIP deployment with firewalls and NATs).

Check these links for some infomation about SIP security. I found them 
helpful.

http://www.dynamicsoft.com/news/presentations/SIPSecurity.pdf

http://www.off-pisteconsulting.com/research/pubs/ndss03-slides.ppt

Considering your question about spontanous calls: I think there are 
several reasons that make this not possible.

Except DOS attacks to SIP servers/user agents, one of the issue that I 
was thinking about is:

you can very easily write a little script that generates INVITE messages 
and sends them to random SIP URLs. Without any authentication 
mechanisms, the result will be: you will make the SIP phones of tousends 
of peaple ringing!!

Note here that this is more a social issue, a sort of SIP spam!!

Any comments are welcome!

If I misunderstood your question, please let me know!

Ali
--
Ali Fessi
NEC Network Laboratories




Khare, Anir wrote:

>Hello,
> 
>Can someone please point me to the known issues with SIP security? 
> 
>Also are there any initiatives in progress for using Liberty alliance
>like identity/trust infrastructure in conjunction with SIP protocols to
>facilitate spontaneous trusted real time communication between a A party
>and a B party, who are not registered with one SIP proxy? 
> 
>Thanks,
>Anir
>
>  
>



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] [Query] SIP security and trusted spontaneous interactions between end points
Date: Mon, 26 Apr 2004 08:46:19 +1000
Message-ID: <1FD836F38652D211A63E0008C7244343125171FE@ntmsg0008.corpmail.telstra.com.au>
Thread-Topic: [Sip] [Query] SIP security and trusted spontaneous interactions between end points
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From: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
To: "Ali Fessi" <ali.fessi@netlab.nec.de>, <sip@ietf.org>
Cc: <jon.peterson@neustar.biz>, <jmpolk@cisco.com>,
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> Except DOS attacks to SIP servers/user agents, one of the issue that I

> was thinking about is:

> you can very easily write a little script that generates INVITE
messages=20
> and sends them to random SIP URLs. Without any authentication=20
> mechanisms, the result will be: you will make the SIP phones of
tousends=20
> of peaple ringing!!

> Note here that this is more a social issue, a sort of SIP spam!!

Yes, this is the scenario I was thinking about.=20

In the existing SIP standard, is there any way for a UAS to specify at
the time of REGISTER or otherwise, to avoid such spams by restricting
the incoming requests to some circle of trust, either introduced by the
SIP proxy server of the domain or by the UAS itself?

Also:
I came across following internet draft draft-ietf-sipping-trait-authz-00
- Trait based Authorization Requirements for Session Initiation Protocol
(SIP) Feb 7, 2004, by Jon Peterson et.al. on SIPPING.

I think, this is a very good step towards handling the spam issues by
tracking the sender's traits.

Liberty alliances deal with such issues in the context of http
interactions between service consumer and service provider of online
(web) services. Along with other standard single sign on scenarios,
liberty also covers the scenarios like service providers using selective
credentials authorizations (e.g. charging)and attributes (e.g. ZIP code)
provided by identity providers and identity services providers, about a
service consumer (principal, in order to provide access to online
services. The service providers don't need to have user accounts on
their sites and users don't need to disclose their identity to such
service providers. It provides the mechanisms for allowing spontaneous
trusted interactions between the liberty enabled entities.

The interactions between SIP UAC and SIP UAS, in large open communities
with few pre-existing relationships, will have similar issues.=20

Probably the liberty framework (Liberty ID-FF and ID-WSF) can be used as
an authorization service as mentioned in section 5 of the abovementioned
draft by Jon Peterson et.al. Additionally use of Liberty architecture
may streamline some of the issues related to management of authorization
assertions carried by the SIP requests, as proposed in this draft. The
assertions may not be required to be physically added in the SIP
requests. The SIP requests can carry Liberty pseudonyms for example and
the authorization assertions can be retrieved by the UASs from the
identity web services, which are supposed to provide access control in
terms of who can access what attributes.

Just trying to understand, whether there is any synergy between the two
architectures or are there any known issues which prevent a protocol
like SIP from using Liberty architecture for achieving desired trust in
open environments.


 Anir

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Subject: RE: [Sip] [Query] SIP security and trusted spontaneous interactions between end points
Date: Mon, 26 Apr 2004 08:47:17 +1000
Message-ID: <1FD836F38652D211A63E0008C7244343125171FF@ntmsg0008.corpmail.telstra.com.au>
Thread-Topic: [Sip] [Query] SIP security and trusted spontaneous interactions between end points
Thread-Index: AcQpFK/mhjLy9pS9QPqdjjA4SFvtjQBWiLVQACoi9NA=
From: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
To: "Harry Behrens" <hb@snom.de>, <sip@ietf.org>
Cc: <knuettel@fokus.fraunhofer.de>,
        "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
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Hello Harry,

This is interesting.

A general query: Did you consider using Liberty alliance like protocols =
for security and accounting interactions between VN and HN. Are there =
any known issues about using Liberty protocols in this context?

A specific question about the requirements (section 2). Is ability for =
Alice to get a SIP URI in VN a mandatory requirement? i.e. why can't =
Alice use Mobile IP and continue using SIP infrastructure at HN? Is it =
because Alice wants to use some resources like PSTN gateway in the VN?=20

Regards,
Anir=20


-----Original Message-----
From: Harry Behrens [mailto:hb@snom.de]=20
Sent: Friday, 23 April 2004 7:07 PM
To: Khare, Anir
Cc: sip@ietf.org; knuettel@fokus.fraunhofer.de
Subject: Re: [Sip] [Query] SIP security and trusted spontaneous =
interactions between end points

Khare, Anir wrote:

> Hello,
>
> =20
>
> Can someone please point me to the known issues with SIP security?
>
> =20
>

> Also are there any initiatives in progress for using Liberty alliance=20
> like identity/trust infrastructure in conjunction with SIP protocols=20
> to facilitate spontaneous trusted real time communication between a A=20
> party and a B party, who are not registered with one SIP proxy?
>
> =20
>

Karsten Kn=FCttel and I have proposed something to that extent in March =
on=20
the SIPPING list,
which uses PKI to allow dynamic agreement on trust and billing=20
relationships.
Maybe it proves useful to you:

Interdomain Trust-Relationships for SIP-based Roaming

This draft describes an authentication mechanism which can be used to=20
enable users of VoIP to globally roam between any number of ITSPs=20
(Internet Telephony Service Providers) without needing to have a prior=20
customer or billing relationship with all of them. This enables=20
profiles and one-line-billing.=20
Providers using this authentication mechanism do neither need full-mesh=20
trust relationships or roaming agreements with every other possible=20
provider nor do they need to rely on a centralized brokerage entity to=20
process calls.=20
The Home Network (HN), which handles all AAA for a user attempting to=20
use an ITSP's service is determined through a unique identifier=20
submitted by the user. This Network Access Identifier [RFC2486]=20
contains information about the trust domain or trust realm the user=20
belongs to. Based on a simple discovery mechanism a provider can=20
establish a trust path to this home network. Once this is established,=20
the provider can authenticate the user and check his credit with the=20
home network. Accounting and rate information will be sent to the home=20
network for mediation and processing.=20

@
http://www.snom.com/download/draft-behrens-sipping-roaming-architecture-0=
1.txt

Regards,

    Harry

> Thanks,
>
> Anir
>


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John,

I agree with you that some clarification would be helpful. I just have a 
few nits with your draft:

Some other potential pieces of state could be mentioned in section 2:

- a body part with a content-disposition of "alert"
- a body part with the content-disposition of "icon"

These have never been very well explained, and have some semantic 
overlap with the Alert-Info and Call-Info header. It may be that the 
best way to deal with them would be to simply remove these content 
dispositions, or document that they must be used together with a 
Content-Id and a reference from an Alert-Info or Call-Info header.

Section 3.1 says "The UPDATE mechanism [2] provides a means for updating 
the session using a 2-message sequence (request/response) during an 
INVITE-initiated dialog." AFAIK it isn't limited to use in 
invite-initiated dialogs. For instance, it could be used for target 
refresh in a subscribe-initiated dialog.

Section 4 makes some specific proposals. One it doesn't make, that 
shoudl be considered, is updating 3261 to further elaborate the state 
that is maintained as part of a dialog, and that is subject to change.

Re the open issue in section 5: presumably the purpose of sending an 
UPDATE is to change some aspect of the state of the dialog. No matter 
what piece of state that is, sending overlapping UPDATE requests makes 
it difficult to reach a consistent agreement about what the state of the 
dialog is. For instance:

	A                   B
	| (1)UPDATE(From=C) |
	|------------------>|
	|                   |
	| (2)UPDATE(From=D) |
	|------------------>|
	|                   |
	| 200 OK (2)        |
	|<------------------|
	| 200 OK (1)        |
	|<------------------|

(Perhaps this happened because message (1) got lost in the network for 
awhile.)

You could try to tighten this up by saying that you can't have 
concurrent requests outstanding that revise the same aspect of state. 
But that starts to get very complex to define precisely, especially 
since some aspects of state (like the Contact) are required to be 
transmitted in every UPDATE or reINVITE, even if unchanged.

But its perhaps simpler to just leave this alone.

	Paul


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Date: Fri, 23 Apr 2004 21:17:15 -1000
Subject: Re: RE : [Sip] History-Info Privacy Issue
From: Cullen Jennings <fluffy@cisco.com>
To: PROUVOST S=?ISO-8859-1?B?6Q==?=bastien FTRD/DAC/ISS
	<sebastien.prouvost@francetelecom.com>,
        "Jesske, R" <R.Jesske@t-com.net>, <mary.barnes@nortelnetworks.com>,
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I think that is a fair concern. The privacy of this is going to be somewhat
confusing at best. Defining it across two drafts will add to the confusion.
I still don=B9t care greatly one way or the other but I see your point.


On 4/22/04 8:14 AM, "PROUVOST S=E9bastien FTRD/DAC/ISS"
<sebastien.prouvost@francetelecom.com> wrote:

> I also support 1 because I think 2 will lead to have two different privac=
y
> mechanisms associated with a single header (depending on whether the
> additional draft is supported or not) which is not what we want.
> =20
> Sebastien Prouvost
> France Telecom R&D
>> =20
>> =20
>> -----Message d'origine-----
>> De : sip-admin@ietf.org  [mailto:sip-admin@ietf.org] De la part de Jessk=
e,  R
>> Envoy=E9 : jeudi 22 avril 2004 15:20
>> =C0 :  fluffy@cisco.com; mary.barnes@nortelnetworks.com;  sip@ietf.org
>> Objet : AW: [Sip] History-Info Privacy  Issue
>>=20
>> =20
>> I  favour 1, because I think this should be kept together.
>> =20
>> =20
>> =20
>> Best  Regards
>> =20
>> =20
>> =20
>> Roland
>> =20
>>> =20
>>> -----Urspr=FCngliche Nachricht-----
>>> Von: sip-admin@ietf.org  [mailto:sip-admin@ietf.org]Im Auftrag von Cull=
en
>>> Jennings
>>> Gesendet: Donnerstag, 22. April 2004 00:04
>>> An:  Mary Barnes; sip@ietf.org
>>> Betreff: Re: [Sip] History-Info Privacy  Issue
>>>=20
>>>=20
>>> I don't care a huge amount but I think I  slightly favor 2.
>>>=20
>>>=20
>>> On 4/16/04 9:05 AM, "Mary Barnes"  <mary.barnes@nortelnetworks.com> wro=
te:
>>>=20
>>> =20
>>>> Per my  posting on the status of History-Info, Privacy is one of the
>>>> remaining  open issues.  The proposal currently in the draft is quite
>>>> simple: =20
>>>>=20
>>>> If the Privacy header has a priv-value of "Session" or "Header"  priva=
cy,
>>>> the History-Info  SHOULD not be included in the Request (or  subsequen=
t
>>>> responses). =20
>>>>=20
>>>> This is effectively applying the  fundamental recommendation in RFC332=
3
>>>> that any optional information that  can divulge information about the
>>>> user(s) should not be included in the  requests.
>>>>=20
>>>> A proposal was made that there are scenarios (e.g.  where local policy
>>>> limits the applicability of History-Info within a  specific domain) wh=
ere
>>>> you would want to capture the History-Info but not  send it beyond cer=
tain
>>>> intermediaries or domains.  To support this,  it was proposed that we =
could
>>>> extend the Privacy header defined in RFC  3323 with tags for History-I=
nfo
>>>> something like the  following:
>>>>=20
>>>> priv-value =3D "history" / "history-index"
>>>>=20
>>>> The "history" priv-value would mean that all the  History-Info entries
>>>> should be removed.  The "history-index" would be  used to indicate tha=
t the
>>>> specific entry is to be removed and would need  to have the Privacy he=
ader
>>>> escaped in the URI so that it can be associated  with a specific
>>>> History-Info entry (or we need to add another field to  History-Info f=
or
>>>> this).   =20
>>>>=20
>>>> So, I think we have two  options here:
>>>> 1) Extend History-Info to have an optional privacy field  along the li=
nes
>>>> of what's detailed above.
>>>>=20
>>>> 2) Defer the  definition of the privacy field to a separate draft, sin=
ce it
>>>> appears to  be primarily limited to scenarios with certain trust model=
s.
>>>>=20
>>>> So, if folks could just take a bit of time and consider  their prefere=
nce,
>>>> I would like to get the draft updated soon and closed  off prior to th=
e
>>>> Interim meeting.
>>>>=20
>>>> Regards,=20
>>>> Mary H. Barnes
>>>> mary.barnes@nortelnetworks.com
>>>>=20
>>>=20
>=20



--B_3165814081_1341657
Content-type: text/html; charset="ISO-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: RE : [Sip] History-Info Privacy Issue</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
I think that is a fair concern. The privacy of this is going to be somewhat=
 confusing at best. Defining it across two drafts will add to the confusion.=
 I still don&#8217;t care greatly one way or the other but I see your point.=
 <BR>
<BR>
<BR>
On 4/22/04 8:14 AM, &quot;PROUVOST S&eacute;bastien FTRD/DAC/ISS&quot; &lt;=
sebastien.prouvost@francetelecom.com&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#0000=
FF"><FONT FACE=3D"Arial">I also support 1 because I think 2 will lead to have =
two different privacy mechanisms associated with a single header (depending =
on whether the additional draft is supported or not) which is not what we wa=
nt.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">Sebastien Prouvost<BR>
France Telecom R&amp;D<BR>
</FONT></FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D=
"Verdana"> <BR>
&nbsp;<BR>
</FONT><FONT FACE=3D"Tahoma">-----Message d'origine-----<BR>
<B>De :</B> sip-admin@ietf.org &nbsp;[<a href=3D"mailto:sip-admin@ietf.org]">=
mailto:sip-admin@ietf.org]</a> <B>De la part de</B> Jesske, &nbsp;R<BR>
<B>Envoy&eacute; :</B> jeudi 22 avril 2004 15:20<BR>
<B>&Agrave; :</B> &nbsp;fluffy@cisco.com; mary.barnes@nortelnetworks.com; &=
nbsp;sip@ietf.org<BR>
<B>Objet :</B> AW: [Sip] History-Info Privacy &nbsp;Issue<BR>
<BR>
</FONT><FONT FACE=3D"Verdana"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">I &nbsp;favour 1, because I=
 think this should be kept together.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"> <BR>
&nbsp;<BR>
&nbsp;<BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">Best &nbsp;Regards<BR>
</FONT></FONT><FONT FACE=3D"Verdana"> <BR>
&nbsp;<BR>
&nbsp;<BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">Roland<BR>
</FONT></FONT><FONT FACE=3D"Verdana"> <BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verdan=
a"> <BR>
</FONT><FONT FACE=3D"Tahoma">-----Urspr&uuml;ngliche Nachricht-----<BR>
<B>Von:</B> sip-admin@ietf.org &nbsp;[<a href=3D"mailto:sip-admin@ietf.org]">=
mailto:sip-admin@ietf.org]</a><B>Im Auftrag von </B>Cullen &nbsp;Jennings<BR=
>
<B>Gesendet:</B> Donnerstag, 22. April 2004 00:04<BR>
<B>An:</B> &nbsp;Mary Barnes; sip@ietf.org<BR>
<B>Betreff:</B> Re: [Sip] History-Info Privacy &nbsp;Issue<BR>
<BR>
</FONT><FONT FACE=3D"Verdana"><BR>
I don't care a huge amount but I think I &nbsp;slightly favor 2. <BR>
<BR>
<BR>
On 4/16/04 9:05 AM, &quot;Mary Barnes&quot; &nbsp;&lt;mary.barnes@nortelnet=
works.com&gt; wrote:<BR>
<BR>
&nbsp;<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verdan=
a">Per my &nbsp;posting on the status of History-Info, Privacy is one of the=
 remaining &nbsp;open issues. &nbsp;The proposal currently in the draft is q=
uite simple: &nbsp;<BR>
<BR>
If the Privacy header has a priv-value of &quot;Session&quot; or &quot;Head=
er&quot; &nbsp;privacy, the History-Info &nbsp;SHOULD not be included in the=
 Request (or &nbsp;subsequent responses). &nbsp;<BR>
<BR>
This is effectively applying the &nbsp;fundamental recommendation in RFC332=
3 that any optional information that &nbsp;can divulge information about the=
 user(s) should not be included in the &nbsp;requests. &nbsp;<BR>
<BR>
A proposal was made that there are scenarios (e.g. &nbsp;where local policy=
 limits the applicability of History-Info within a &nbsp;specific domain) wh=
ere you would want to capture the History-Info but not &nbsp;send it beyond =
certain intermediaries or domains. &nbsp;To support this, &nbsp;it was propo=
sed that we could extend the Privacy header defined in RFC &nbsp;3323 with t=
ags for History-Info something like the &nbsp;following:<BR>
<BR>
priv-value =3D &quot;history&quot; / &quot;history-index&quot; &nbsp;&nbsp;<B=
R>
<BR>
The &quot;history&quot; priv-value would mean that all the &nbsp;History-In=
fo entries should be removed. &nbsp;The &quot;history-index&quot; would be &=
nbsp;used to indicate that the specific entry is to be removed and would nee=
d &nbsp;to have the Privacy header escaped in the URI so that it can be asso=
ciated &nbsp;with a specific History-Info entry (or we need to add another f=
ield to &nbsp;History-Info for this). &nbsp;&nbsp;&nbsp;<BR>
<BR>
So, I think we have two &nbsp;options here: <BR>
1) Extend History-Info to have an optional privacy field &nbsp;along the li=
nes of what's detailed above. &nbsp;<BR>
<BR>
2) Defer the &nbsp;definition of the privacy field to a separate draft, sin=
ce it appears to &nbsp;be primarily limited to scenarios with certain trust =
models. &nbsp;&nbsp;<BR>
<BR>
So, if folks could just take a bit of time and consider &nbsp;their prefere=
nce, I would like to get the draft updated soon and closed &nbsp;off prior t=
o the Interim meeting.<BR>
<BR>
Regards, <BR>
Mary H. Barnes &nbsp;<BR>
mary.barnes@nortelnetworks.com <BR>
<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verda=
na"><BR>
</FONT></SPAN></BLOCKQUOTE></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FON=
T FACE=3D"Verdana"><BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verda=
na"><BR>
</FONT></SPAN>
</BODY>
</HTML>


--B_3165814081_1341657--


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Thread-Topic: Application identification with callee caps/caller preferences
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Hi, I have a question on following scenario.=20

The user Y has two games installed Chess (named x-chess) and Othello =
(named x-othello). The caller X has the equivalent chess-game (x-chess) =
and would like to use SIP to invite Y for a game.=20
Is the following approach correct according to the callee caps/caller =
preferences specs? Namely is it correct to use the 'type' media feature =
tag in this way. In practice here the 'type' tag identifies the =
application and on the other hand defines also the media capability of =
the UA (since the x-chess in this case might use whatever =
"gaming-protocol" as media stream).

The callee Y would generate a registration which looks like, in part:

For the Chess game:

REGISTER sip:example.com SIP/2.0
   To: sip:Y@example.com
   Contact: sip:Y1@pc.example.com
     ;type=3D"application/x-chess"

For the Othello game:

REGISTER sip:example.com SIP/2.0
   To: sip:Y@example.com
   Contact: sip:Y2@pc.example.com
     ;type=3D"application/x-othello"

Now when the caller X wants to play x-chess it would express its' =
preference by generating an INVITE which looks like, in part:

INVITE sip:Y@example.com SIP/2.0
   From: sip:X@example.com
   To: sip:Y@example.com
   Accept-Contact: *;type=3D"application/x-chess";require;explicit


Best Regards,
Jari

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Date: Tue, 27 Apr 2004 09:48:22 -0400
From: David Oran <oran@cisco.com>
To: Jari.P.Kinnunen@nokia.com, sip@ietf.org
Subject: Re: [Sip] Application identification with callee caps/caller
 preferences
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 >
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This seems a bit of a stretch for callee caps. What bothers me is the 
notion i that you want to play a game with somebody, and which game is 
determined by a preference matching scheme. I suppose there's nothing 
horrible in that - it just strikes me as weird.

I would think that you would be more likely to choose the combination of 
participant+game through addressing, such as by having multiple 
registrations of the form

sip:Y.othello@example.com and sip:Y.chess@example.com

This has the additional benefit of not requiring a registry for game names.

Dave.

--On Tuesday, April 27, 2004 10:38 AM +0300 Jari.P.Kinnunen@nokia.com wrote:

> Hi, I have a question on following scenario.
>
> The user Y has two games installed Chess (named x-chess) and Othello
> (named x-othello). The caller X has the equivalent chess-game (x-chess)
> and would like to use SIP to invite Y for a game.  Is the following
> approach correct according to the callee caps/caller preferences specs?
> Namely is it correct to use the 'type' media feature tag in this way. In
> practice here the 'type' tag identifies the application and on the other
> hand defines also the media capability of the UA (since the x-chess in
> this case might use whatever "gaming-protocol" as media stream).
>
> The callee Y would generate a registration which looks like, in part:
>
> For the Chess game:
>
> REGISTER sip:example.com SIP/2.0
>    To: sip:Y@example.com
>    Contact: sip:Y1@pc.example.com
>      ;type="application/x-chess"
>
> For the Othello game:
>
> REGISTER sip:example.com SIP/2.0
>    To: sip:Y@example.com
>    Contact: sip:Y2@pc.example.com
>      ;type="application/x-othello"
>
> Now when the caller X wants to play x-chess it would express its'
> preference by generating an INVITE which looks like, in part:
>
> INVITE sip:Y@example.com SIP/2.0
>    From: sip:X@example.com
>    To: sip:Y@example.com
>    Accept-Contact: *;type="application/x-chess";require;explicit
>
>
> Best Regards,
> Jari
>
> _______________________________________________
> 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: David Oran <oran@cisco.com>
CC: Jari.P.Kinnunen@nokia.com, sip@ietf.org
Subject: Re: [Sip] Application identification with callee caps/caller preferences
References: <0C1353ABB1DEB74DB067ADFF749C4EEF029E151D@esebe004.ntc.nokia.com > <F586CA84FE30BD374874DF77@[10.32.245.156]>
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I think what is proposed is quite reasonable. Given that the game is 
defined by the type of media stream used to interact, the simplest 
approach is simply to send an INVITE to the AoR, offering the desired 
kind of media stream and not bothering with callerprefs at all. In that 
case the invite would be forked to all the callee's devices. Presumably 
those that couldn't handle the media type would refuse the call, and any 
that so support it and are available might respond with a 180.

Using callerprefs here is possibly a slight optimization, especially in 
case the proxy were to try serial forking. And it is also insurance 
against devices that might accept the call while refusing the media.

This would be more important if you want two streams - one for the game 
and another for voice. If you simply offer both then you run a real risk 
that an audio phone might accept the call and refuse the game stream.

Of course it is also possible to use separate addresses for the games. 
But that might be inconvenient, and limiting if the game console also 
does voice and/or video.

	Paul

David Oran wrote:
> This seems a bit of a stretch for callee caps. What bothers me is the 
> notion i that you want to play a game with somebody, and which game is 
> determined by a preference matching scheme. I suppose there's nothing 
> horrible in that - it just strikes me as weird.
> 
> I would think that you would be more likely to choose the combination of 
> participant+game through addressing, such as by having multiple 
> registrations of the form
> 
> sip:Y.othello@example.com and sip:Y.chess@example.com
> 
> This has the additional benefit of not requiring a registry for game names.
> 
> Dave.
> 
> --On Tuesday, April 27, 2004 10:38 AM +0300 Jari.P.Kinnunen@nokia.com 
> wrote:
> 
>> Hi, I have a question on following scenario.
>>
>> The user Y has two games installed Chess (named x-chess) and Othello
>> (named x-othello). The caller X has the equivalent chess-game (x-chess)
>> and would like to use SIP to invite Y for a game.  Is the following
>> approach correct according to the callee caps/caller preferences specs?
>> Namely is it correct to use the 'type' media feature tag in this way. In
>> practice here the 'type' tag identifies the application and on the other
>> hand defines also the media capability of the UA (since the x-chess in
>> this case might use whatever "gaming-protocol" as media stream).
>>
>> The callee Y would generate a registration which looks like, in part:
>>
>> For the Chess game:
>>
>> REGISTER sip:example.com SIP/2.0
>>    To: sip:Y@example.com
>>    Contact: sip:Y1@pc.example.com
>>      ;type="application/x-chess"
>>
>> For the Othello game:
>>
>> REGISTER sip:example.com SIP/2.0
>>    To: sip:Y@example.com
>>    Contact: sip:Y2@pc.example.com
>>      ;type="application/x-othello"
>>
>> Now when the caller X wants to play x-chess it would express its'
>> preference by generating an INVITE which looks like, in part:
>>
>> INVITE sip:Y@example.com SIP/2.0
>>    From: sip:X@example.com
>>    To: sip:Y@example.com
>>    Accept-Contact: *;type="application/x-chess";require;explicit
>>
>>
>> Best Regards,
>> Jari
>>
>> _______________________________________________
>> 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: David Oran <oran@cisco.com>
CC: Jari.P.Kinnunen@nokia.com, sip@ietf.org
Subject: Re: [Sip] Application identification with callee caps/caller preferences
References: <0C1353ABB1DEB74DB067ADFF749C4EEF029E151D@esebe004.ntc.nokia.com > <F586CA84FE30BD374874DF77@[10.32.245.156]>
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David Oran wrote:
> sip:Y.othello@example.com and sip:Y.chess@example.com
> 
> This has the additional benefit of not requiring a registry for game names.

This alternative has the additional disadvantage of either not resolving 
through any proxy/registrar that I know of, or of requiring a seperate 
REGISTER message and authentication transacation for each client-game 
combination (not to mention account creation). Suppose I had three 
hundred games on my PDA . . .

Use of a preference/capability tag for a "chess" call gets routed to a 
machine with that game is just as reasonable as using pref/cap matching 
to make sure a "video" call request gets routed to a display device with 
a camera and sound.

Of course, I'm still deeply confused between:
1) preference/capability negotiation enabled by offer/answer
2) preference/capability negotiation enabled by callee-caps/caller-pref.


The issue of having a huge registry of names seems to be valid, however.

Options that come to mind:

1) Non-registered "user capability" sub-tags
2) More SDP extensions . . . we're really describing the session we 
want, not the capabilities of the target.
3) URI parameters, anybody?

--
Dean

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Subject: RE: [Sip] Application identification with callee caps/caller preferences
Date: Tue, 27 Apr 2004 13:07:09 -0400
Message-ID: <34DA635B184A644DA4588E260EC0A25A06E667C7@ACCLUST02EVS1.ugd.att.com>
Thread-Topic: [Sip] Application identification with callee caps/caller preferences
Thread-Index: AcQsdAwmjBEEZh7HRtmpqbeO/INVqgABU04w
From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "David Oran" <oran@cisco.com>
Cc: <Jari.P.Kinnunen@nokia.com>, <sip@ietf.org>
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Is not that game will be a kind of "shared application" consisting of
audio synchronized with graphics? If so, SDP itself may also provide an
opportunity for negotiations.

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Paul
Kyzivat
Sent: Tuesday, April 27, 2004 11:51 AM
To: David Oran
Cc: Jari.P.Kinnunen@nokia.com; sip@ietf.org
Subject: Re: [Sip] Application identification with callee caps/caller
preferences


I think what is proposed is quite reasonable. Given that the game is=20
defined by the type of media stream used to interact, the simplest=20
approach is simply to send an INVITE to the AoR, offering the desired=20
kind of media stream and not bothering with callerprefs at all. In that=20
case the invite would be forked to all the callee's devices. Presumably=20
those that couldn't handle the media type would refuse the call, and any

that so support it and are available might respond with a 180.

Using callerprefs here is possibly a slight optimization, especially in=20
case the proxy were to try serial forking. And it is also insurance=20
against devices that might accept the call while refusing the media.

This would be more important if you want two streams - one for the game=20
and another for voice. If you simply offer both then you run a real risk

that an audio phone might accept the call and refuse the game stream.

Of course it is also possible to use separate addresses for the games.=20
But that might be inconvenient, and limiting if the game console also=20
does voice and/or video.

	Paul

David Oran wrote:
> This seems a bit of a stretch for callee caps. What bothers me is the=20
> notion i that you want to play a game with somebody, and which game is

> determined by a preference matching scheme. I suppose there's nothing=20
> horrible in that - it just strikes me as weird.
>=20
> I would think that you would be more likely to choose the combination
of=20
> participant+game through addressing, such as by having multiple=20
> registrations of the form
>=20
> sip:Y.othello@example.com and sip:Y.chess@example.com
>=20
> This has the additional benefit of not requiring a registry for game
names.
>=20
> Dave.
>=20
> --On Tuesday, April 27, 2004 10:38 AM +0300 Jari.P.Kinnunen@nokia.com=20
> wrote:
>=20
>> Hi, I have a question on following scenario.
>>
>> The user Y has two games installed Chess (named x-chess) and Othello
>> (named x-othello). The caller X has the equivalent chess-game
(x-chess)
>> and would like to use SIP to invite Y for a game.  Is the following
>> approach correct according to the callee caps/caller preferences
specs?
>> Namely is it correct to use the 'type' media feature tag in this way.
In
>> practice here the 'type' tag identifies the application and on the
other
>> hand defines also the media capability of the UA (since the x-chess
in
>> this case might use whatever "gaming-protocol" as media stream).
>>
>> The callee Y would generate a registration which looks like, in part:
>>
>> For the Chess game:
>>
>> REGISTER sip:example.com SIP/2.0
>>    To: sip:Y@example.com
>>    Contact: sip:Y1@pc.example.com
>>      ;type=3D"application/x-chess"
>>
>> For the Othello game:
>>
>> REGISTER sip:example.com SIP/2.0
>>    To: sip:Y@example.com
>>    Contact: sip:Y2@pc.example.com
>>      ;type=3D"application/x-othello"
>>
>> Now when the caller X wants to play x-chess it would express its'
>> preference by generating an INVITE which looks like, in part:
>>
>> INVITE sip:Y@example.com SIP/2.0
>>    From: sip:X@example.com
>>    To: sip:Y@example.com
>>    Accept-Contact: *;type=3D"application/x-chess";require;explicit
>>
>>
>> Best Regards,
>> Jari
>>
>> _______________________________________________
>> 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
>=20
>=20
>=20
>=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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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
Use sipping@ietf.org for new developments on the application of sip



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To: Dean Willis <dean.willis@softarmor.com>
CC: David Oran <oran@cisco.com>, Jari.P.Kinnunen@nokia.com, sip@ietf.org
Subject: Re: [Sip] Application identification with callee caps/caller preferences
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Dean Willis wrote:
> David Oran wrote:
> 
>> sip:Y.othello@example.com and sip:Y.chess@example.com
>>
>> This has the additional benefit of not requiring a registry for game 
>> names.
> 
> 
> This alternative has the additional disadvantage of either not resolving 
> through any proxy/registrar that I know of, or of requiring a seperate 
> REGISTER message and authentication transacation for each client-game 
> combination (not to mention account creation). Suppose I had three 
> hundred games on my PDA . . .
> 
> Use of a preference/capability tag for a "chess" call gets routed to a 
> machine with that game is just as reasonable as using pref/cap matching 
> to make sure a "video" call request gets routed to a display device with 
> a camera and sound.
> 
> Of course, I'm still deeply confused between:
> 1) preference/capability negotiation enabled by offer/answer
> 2) preference/capability negotiation enabled by callee-caps/caller-pref.

See my earlier reply.

Forking and offer/answer provide a partial solution. But offer/answer 
doesn't give you a way to say you don't want the call unless your 
offered media are accepted. At least not an easy way. Conceivably some 
new extension to preconditions could be made to handle this kind of 
thing. Lacking that you run the risk of having an unsuitable device 
answer, even if there is a suitable device present.

Callerprefs gives you an increased likelihood that you will end up with 
a suitable device.

> The issue of having a huge registry of names seems to be valid, however.

I don't see how you can avoid that, one way or another. If you are going 
to set up the session with sip, then there needs to be some media 
stream. If the media stream protocol is game dependent then it needs to 
be registered somehow to be referenced in an m-line.

I suppose the alternative is some generic game protocol. But I doubt 
that would fly, and even if it did, you would still have to discover 
somehow what games you had in common, and pick one.

> Options that come to mind:
> 
> 1) Non-registered "user capability" sub-tags
> 2) More SDP extensions . . . we're really describing the session we 
> want, not the capabilities of the target.
> 3) URI parameters, anybody?

These really are capabilities. But you don't necessarily always enable 
all your capabilities, and so perhaps don't offer them all by default. 
This is a relatively new concept in practice, since practically speaking 
most sip devices today only do voice. (I'll bet that video phones aren't 
always going to want to offer video - there will probably be a button 
that turns that on/off.)

If there are a *lot* of supported game protocols, and you just want to 
dynamically choose one that both ends support, then I think you will 
need a separate way of negotiating that. Maybe you simply establish an 
MSRP IM session, discuss what game to play, and then reinvite with an 
offer that adds a stream for the chosen game. Of course, if you started 
out with IM, the other party might not be IMing from a game console. But 
that should simply be a matter of him first transferring his end of the 
call to the game console before gaming begins. E.g.

- Alice, from game console, opens IM session with Bob
- IM session is established on Bob's PC.
- Alice sends message to Bob espressing an interest in playing game.
- They discuss what game to play. Decide on Doom.
- Bob transfers the IM session to his X-Box so he has the good
   audio, TV screen, and game controllers. (This could also be
   done by conferencing in the X-Box.)
- Alice sends reinvite, adding a Doom media session. This goes
   to Bob's X-Box. Now have both Doom and IM.
- They play ...

Other possibilities:

- simply send invite with a separate m-line for each supported game, 
using ALT to say they are mutually exclusive.

- publish presence, listing all the types (including games) supported. 
Then watcher can choose which of those to offer when calling.

	Paul


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From exim@www1.ietf.org  Wed Apr 28 12:07:37 2004
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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: SIP WG <sip@ietf.org>
Cc: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
Date: Wed, 28 Apr 2004 16:24:06 +0100
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Subject: [Sip] RFC 3261 bug in description of 487?
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It seems that the description of a 487 response in RFC 3261 is incomplete.
Consider the following sections from RFC 3261.

21.4.25 487 Request Terminated

   The request was terminated by a BYE or CANCEL request.  This response
   is never returned for a CANCEL request itself.

This might be interpreted as meaning that a 487 should _only_ be sent if
a request has been terminated by a BYE or a CANCEL.  However this is
not consistent with other parts of RFC 3261, where a 487 is generated
without the involvement of either BYE or CANCEL.  One example of this
is section 13.3.1, reproduced here.

13.3.1 Processing of the INVITE

   The UAS core will receive INVITE requests from the transaction layer.
   It first performs the request processing procedures of Section 8.2,
   which are applied for both requests inside and outside of a dialog.

   Assuming these processing states are completed without generating a
   response, the UAS core performs the additional processing steps:

      1. If the request is an INVITE that contains an Expires header
         field, the UAS core sets a timer for the number of seconds
         indicated in the header field value.  When the timer fires, the
         invitation is considered to be expired.  If the invitation
         expires before the UAS has generated a final response, a 487
         (Request Terminated) response SHOULD be generated.

In the this case a 487 response is generated without the involvement of
either CANCEL or BYE.

Am I correct in believing that sections such as 13.3.1 are correct and
there is simply a bug in that section 21.4.25 is incomplete?

Regards,
Paul DS

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


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Subject: RE : [Sip] History-Info Privacy Issue
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From: =?iso-8859-1?Q?PROUVOST_S=E9bastien_FTRD/DAC/ISS?= <sebastien.prouvost@francetelecom.com>
To: "Jesske, R" <R.Jesske@t-com.net>, <fluffy@cisco.com>,
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I also support 1 because I think 2 will lead to have two different =
privacy mechanisms associated with a single header (depending on whether =
the additional draft is supported or not) which is not what we want.
=20
Sebastien Prouvost
France Telecom R&D

-----Message d'origine-----
De : sip-admin@ietf.org [mailto:sip-admin@ietf.org] De la part de =
Jesske, R
Envoy=E9 : jeudi 22 avril 2004 15:20
=C0 : fluffy@cisco.com; mary.barnes@nortelnetworks.com; sip@ietf.org
Objet : AW: [Sip] History-Info Privacy Issue


I favour 1, because I think this should be kept together.
=20
Best Regards
=20
Roland

-----Urspr=FCngliche Nachricht-----
Von: sip-admin@ietf.org [mailto:sip-admin@ietf.org]Im Auftrag von Cullen =
Jennings
Gesendet: Donnerstag, 22. April 2004 00:04
An: Mary Barnes; sip@ietf.org
Betreff: Re: [Sip] History-Info Privacy Issue



I don't care a huge amount but I think I slightly favor 2.=20


On 4/16/04 9:05 AM, "Mary Barnes" <mary.barnes@nortelnetworks.com> =
wrote:



Per my posting on the status of History-Info, Privacy is one of the =
remaining open issues.  The proposal currently in the draft is quite =
simple:=20

If the Privacy header has a priv-value of "Session" or "Header" privacy, =
the History-Info  SHOULD not be included in the Request (or subsequent =
responses). =20

This is effectively applying the fundamental recommendation in RFC3323 =
that any optional information that can divulge information about the =
user(s) should not be included in the requests. =20

A proposal was made that there are scenarios (e.g. where local policy =
limits the applicability of History-Info within a specific domain) where =
you would want to capture the History-Info but not send it beyond =
certain intermediaries or domains.  To support this, it was proposed =
that we could extend the Privacy header defined in RFC 3323 with tags =
for History-Info something like the following:

priv-value =3D "history" / "history-index" =20

The "history" priv-value would mean that all the History-Info entries =
should be removed.  The "history-index" would be used to indicate that =
the specific entry is to be removed and would need to have the Privacy =
header escaped in the URI so that it can be associated with a specific =
History-Info entry (or we need to add another field to History-Info for =
this).   =20

So, I think we have two options here:=20
1) Extend History-Info to have an optional privacy field along the lines =
of what's detailed above. =20

2) Defer the definition of the privacy field to a separate draft, since =
it appears to be primarily limited to scenarios with certain trust =
models. =20

So, if folks could just take a bit of time and consider their =
preference, I would like to get the draft updated soon and closed off =
prior to the Interim meeting.

Regards,=20
Mary H. Barnes=20
mary.barnes@nortelnetworks.com=20







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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D138164014-22042004><FONT face=3DArial color=3D#0000ff =
size=3D2>I also=20
support 1 because I think 2 will lead to have two different privacy =
mechanisms=20
associated with a single header (depending on whether the additional =
draft is=20
supported or not) which is not what we want.</FONT></SPAN></DIV>
<DIV><SPAN class=3D138164014-22042004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D138164014-22042004><SPAN =
class=3D627211215-22042004><FONT=20
face=3DArial color=3D#0000ff size=3D2>Sebastien =
Prouvost</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D138164014-22042004><SPAN =
class=3D627211215-22042004><FONT=20
face=3DArial color=3D#0000ff size=3D2>France Telecom=20
R&amp;D</FONT></SPAN></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr =
align=3Dleft><FONT face=3DTahoma=20
  size=3D2>-----Message d'origine-----<BR><B>De&nbsp;:</B> =
sip-admin@ietf.org=20
  [mailto:sip-admin@ietf.org] <B>De la part de</B> Jesske,=20
  R<BR><B>Envoy=E9&nbsp;:</B> jeudi 22 avril 2004 =
15:20<BR><B>=C0&nbsp;:</B>=20
  fluffy@cisco.com; mary.barnes@nortelnetworks.com;=20
  sip@ietf.org<BR><B>Objet&nbsp;:</B> AW: [Sip] History-Info Privacy=20
  Issue<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
  favour 1, because I think this should be kept =
together.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff size=3D2>Best=20
  Regards</FONT></SPAN></DIV>
  <DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D745261713-22042004><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Roland</FONT></SPAN></DIV>
  <BLOCKQUOTE>
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Urspr=FCngliche Nachricht-----<BR><B>Von:</B> =
sip-admin@ietf.org=20
    [mailto:sip-admin@ietf.org]<B>Im Auftrag von </B>Cullen=20
    Jennings<BR><B>Gesendet:</B> Donnerstag, 22. April 2004 =
00:04<BR><B>An:</B>=20
    Mary Barnes; sip@ietf.org<BR><B>Betreff:</B> Re: [Sip] History-Info =
Privacy=20
    Issue<BR><BR></FONT></DIV><FONT face=3DVerdana><SPAN=20
    style=3D"FONT-SIZE: 14px"><BR>I don't care a huge amount but I think =
I=20
    slightly favor 2. <BR><BR><BR>On 4/16/04 9:05 AM, "Mary Barnes"=20
    &lt;mary.barnes@nortelnetworks.com&gt; wrote:<BR><BR></SPAN></FONT>
    <BLOCKQUOTE><FONT face=3DVerdana><SPAN style=3D"FONT-SIZE: 14px">Per =
my=20
      posting on the status of History-Info, Privacy is one of the =
remaining=20
      open issues. &nbsp;The proposal currently in the draft is quite =
simple:=20
      <BR><BR>If the Privacy header has a priv-value of "Session" or =
"Header"=20
      privacy, the History-Info &nbsp;SHOULD not be included in the =
Request (or=20
      subsequent responses). &nbsp;<BR><BR>This is effectively applying =
the=20
      fundamental recommendation in RFC3323 that any optional =
information that=20
      can divulge information about the user(s) should not be included =
in the=20
      requests. &nbsp;<BR><BR>A proposal was made that there are =
scenarios (e.g.=20
      where local policy limits the applicability of History-Info within =
a=20
      specific domain) where you would want to capture the History-Info =
but not=20
      send it beyond certain intermediaries or domains. &nbsp;To support =
this,=20
      it was proposed that we could extend the Privacy header defined in =
RFC=20
      3323 with tags for History-Info something like the=20
      following:<BR><BR>priv-value =3D "history" / "history-index"=20
      &nbsp;<BR><BR>The "history" priv-value would mean that all the=20
      History-Info entries should be removed. &nbsp;The "history-index" =
would be=20
      used to indicate that the specific entry is to be removed and =
would need=20
      to have the Privacy header escaped in the URI so that it can be =
associated=20
      with a specific History-Info entry (or we need to add another =
field to=20
      History-Info for this). &nbsp;&nbsp;&nbsp;<BR><BR>So, I think we =
have two=20
      options here: <BR>1) Extend History-Info to have an optional =
privacy field=20
      along the lines of what's detailed above. &nbsp;<BR><BR>2) Defer =
the=20
      definition of the privacy field to a separate draft, since it =
appears to=20
      be primarily limited to scenarios with certain trust models.=20
      &nbsp;<BR><BR>So, if folks could just take a bit of time and =
consider=20
      their preference, I would like to get the draft updated soon and =
closed=20
      off prior to the Interim meeting.<BR><BR>Regards, <BR>Mary H. =
Barnes=20
      <BR>mary.barnes@nortelnetworks.com =
<BR><BR></SPAN></FONT></BLOCKQUOTE><FONT=20
    face=3DVerdana><SPAN=20
style=3D"FONT-SIZE: =
14px"><BR></BLOCKQUOTE></BLOCKQUOTE></SPAN></FONT></BODY></HTML>

------_=_NextPart_001_01C4287C.7B776D2D--

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From: Harry Behrens <hb@snom.de>
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CC: sip@ietf.org, knuettel@fokus.fraunhofer.de
Subject: Re: [Sip] [Query] SIP security and trusted spontaneous interactions
 between end points
References: <1FD836F38652D211A63E0008C7244343125171F1@ntmsg0008.corpmail.telstra.com.au>
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Khare, Anir wrote:

> Hello,
>
> =20
>
> Can someone please point me to the known issues with SIP security?
>
> =20
>

> Also are there any initiatives in progress for using Liberty alliance=20
> like identity/trust infrastructure in conjunction with SIP protocols=20
> to facilitate spontaneous trusted real time communication between a A=20
> party and a B party, who are not registered with one SIP proxy?
>
> =20
>

Karsten Kn=FCttel and I have proposed something to that extent in March o=
n=20
the SIPPING list,
which uses PKI to allow dynamic agreement on trust and billing=20
relationships.
Maybe it proves useful to you:

Interdomain Trust-Relationships for SIP-based Roaming

This draft describes an authentication mechanism which can be used to=20
enable users of VoIP to globally roam between any number of ITSPs=20
(Internet Telephony Service Providers) without needing to have a prior=20
customer or billing relationship with all of them. This enables=20
profiles and one-line-billing.=20
Providers using this authentication mechanism do neither need full-mesh=20
trust relationships or roaming agreements with every other possible=20
provider nor do they need to rely on a centralized brokerage entity to=20
process calls.=20
The Home Network (HN), which handles all AAA for a user attempting to=20
use an ITSP's service is determined through a unique identifier=20
submitted by the user. This Network Access Identifier [RFC2486]=20
contains information about the trust domain or trust realm the user=20
belongs to. Based on a simple discovery mechanism a provider can=20
establish a trust path to this home network. Once this is established,=20
the provider can authenticate the user and check his credit with the=20
home network. Accounting and rate information will be sent to the home=20
network for mediation and processing.=20

@
http://www.snom.com/download/draft-behrens-sipping-roaming-architecture-0=
1.txt

Regards,

    Harry

> Thanks,
>
> Anir
>


_______________________________________________
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: "Khare, Anir" <Aniruddha.Khare@team.telstra.com>
CC: sip@ietf.org, knuettel@fokus.fraunhofer.de
Subject: Re: [Sip] [Query] SIP security and trusted spontaneous interactions
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Hello Aniruddha,

inline

Khare, Anir wrote:

>Hello Harry,
>
>This is interesting.
>
>A general query: Did you consider using Liberty alliance like protocols =
for security and accounting interactions between VN and HN. Are there any=
 known issues about using Liberty protocols in this context?
> =20
>

Yes. This was one of our first choices. The problem is that we are=20
trying to be pragmatic (IETF is not a scientists' but an engineers' forum=
).
And the actual usage and implementation of LA is less than assuring of=20
its success and potential ubiquity.
By using a derivate of Diffie-Helman - essentially that is what it boils=20
down to - we kept it generic as well as simple.
Porting this to LA should be straightforward.

>A specific question about the requirements (section 2). Is ability for A=
lice to get a SIP URI in VN a mandatory requirement? i.e. why can't Alice=
 use Mobile IP and continue using SIP infrastructure at HN? Is it because=
 Alice wants to use some resources like PSTN gateway in the VN?=20
> =20
>

That is exactly the point. One can assume that the choice given by=20
"local" carriers to be more competitive in terms of call termination to=20
"local" destinations.
On the other hand, pls. note that we explicitly propose a double=20
registration in section 7.2.2 of the draft.
This assures that Alice can make full use of any services its HN can=20
provide, especially being reachable under her "home" number.
Pls. note that MIP would only become necessary in case of physical=20
handover during dialog.
The scenario is written for "semi-stationary" or "nomadically mobile"=20
(nomadic in the sense that: "users have to shut down an application or a=20
session
and restart it when they connect at the new point of attachment.")
-> Layer-7 mobility
Extensions would have to be used to provide mobility at layer 4 downwards.

Regards,

    Harry


>Regards,
>Anir=20
>
>
>-----Original Message-----
>From: Harry Behrens [mailto:hb@snom.de]=20
>Sent: Friday, 23 April 2004 7:07 PM
>To: Khare, Anir
>Cc: sip@ietf.org; knuettel@fokus.fraunhofer.de
>Subject: Re: [Sip] [Query] SIP security and trusted spontaneous interact=
ions between end points
>
>Khare, Anir wrote:
>
> =20
>
>>Hello,
>>
>>=20
>>
>>Can someone please point me to the known issues with SIP security?
>>
>>=20
>>
>>   =20
>>
>
> =20
>
>>Also are there any initiatives in progress for using Liberty alliance=20
>>like identity/trust infrastructure in conjunction with SIP protocols=20
>>to facilitate spontaneous trusted real time communication between a A=20
>>party and a B party, who are not registered with one SIP proxy?
>>
>>=20
>>
>>   =20
>>
>
>Karsten Kn=FCttel and I have proposed something to that extent in March =
on=20
>the SIPPING list,
>which uses PKI to allow dynamic agreement on trust and billing=20
>relationships.
>Maybe it proves useful to you:
>
>Interdomain Trust-Relationships for SIP-based Roaming
>
>This draft describes an authentication mechanism which can be used to=20
>enable users of VoIP to globally roam between any number of ITSPs=20
>(Internet Telephony Service Providers) without needing to have a prior=20
>customer or billing relationship with all of them. This enables=20
>profiles and one-line-billing.=20
>Providers using this authentication mechanism do neither need full-mesh=20
>trust relationships or roaming agreements with every other possible=20
>provider nor do they need to rely on a centralized brokerage entity to=20
>process calls.=20
>The Home Network (HN), which handles all AAA for a user attempting to=20
>use an ITSP's service is determined through a unique identifier=20
>submitted by the user. This Network Access Identifier [RFC2486]=20
>contains information about the trust domain or trust realm the user=20
>belongs to. Based on a simple discovery mechanism a provider can=20
>establish a trust path to this home network. Once this is established,=20
>the provider can authenticate the user and check his credit with the=20
>home network. Accounting and rate information will be sent to the home=20
>network for mediation and processing.=20
>
>@
>http://www.snom.com/download/draft-behrens-sipping-roaming-architecture-=
01.txt
>
>Regards,
>
>    Harry
>
> =20
>
>>Thanks,
>>
>>Anir
>>
>>   =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



From exim@www1.ietf.org  Thu Apr 29 15:52:44 2004
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Subject: RE: [Sip] Status of SIP MIB
Date: Thu, 29 Apr 2004 13:06:16 -0600
Message-ID: <CD6CE349CFD30D40BF5E13B3E0D8480406A1B3@srvxchg.cablelabs.com>
Thread-Topic: [Sip] Status of SIP MIB
Thread-Index: AcQSyY4uRxGInwV3TvSDaU6mOOII0QbUWE/A
From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Bob Natale" <bnatale@MEGISTO.com>, <Deepak.Pant@utstar.com>
Cc: <sip@ietf.org>, "Kevin Lingle" <klingle@cisco.com>
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Here's one area where you & the wg participants may want to help.

One of the comments we got was:=20
  is the information model we have in the sip mib 07 sufficient to
configure sip *protocol* parameters on basic sip entities (UA, proxy,
registrar, etc.)?

Both kevin and i implemented earlier versions of this mib on some early
sip rfc2543 boxes but speaking for myself, it's been a while I looked
closely at sip devices & their configuration. Back then, I was
considered a fool to push for sip over tcp... so things have changed.

It would be great if some of you could screen sip implementations with
the above question in mind and drop us a minimal set of params you need
to configure *basic* sip entities in a plain text email (no mib syntax
please, just plain text param names with a description and may be a sip
rfc section number as a ref but we can take care of the later). And we
don't care whether those configuration params are provided on a
java-based GUI on a sip phone or via CLI telnet/ssh to a sip proxy or
via some kind of xml/netconf: what matters is the what (what protocol
parameters get configured), not the how (not the actual configuration
protocol & associated data model).

This would be of good help.
Jean-francois.
 jfm@cablelabs.com

> -----Original Message-----
> From: Kevin Lingle [mailto:klingle@cisco.com]=20
> Sent: Thursday, March 25, 2004 4:31 PM
> To: Bob Natale
> Cc: sip@ietf.org
> Subject: Re: [Sip] Status of SIP MIB
>=20
>=20
> bob,
>=20
> you are welcome to review -07 and comment just as anyone
> else is.   we have a set of "official" reviewers (4-5 iirc)
> from which we've fielded 90+ individual comments.  we've=20
> addressed (as in read, corresponded about, given some thought=20
> to, and have a response/solution) a majority of those=20
> comments. the solution - in terms of changes to the mib is=20
> not done in many cases (minor changes are typically done, but=20
> others are larger changes across the mib). as with most=20
> things, it's just a matter of finding the time to work these=20
> changes and work through the remaining comments not yet addressed.
>=20
> on one hand i'd love more review of the draft.
> on the other hand more comments to address will delay
> things more (potentially).
>=20
> considering that i want -08 to be the wglc approved
> revision we take to iesg, i'd say you should review
> -07.
>=20
> do you ever get involved as a mib
> doctor type or expert reviewer of mib drafts that
> have been submitted to iesg?
>=20
> kevin
>=20
>=20
>=20
> Bob Natale wrote:
> > Hi Kevin,
> >=20
> > Thanks...I understand...didn't mean to put more pressure on=20
> you or the=20
> > other authors personally...I'd like to help (I was until=20
> recently in a=20
> > role that kept me from being involved in these=20
> efforts)...would it be=20
> > helpful for me to review the -07 version at this point or should I=20
> > wait for the -08 version?  (Or "neither"! :-)
> >=20
> > Cheers,
> > BobN
> >=20
> >=20
> >>-----Original Message-----
> >>From: Kevin Lingle [mailto:klingle@cisco.com]
> >>Sent: Thursday, March 25, 2004 1:55 PM
> >>To: Bob Natale
> >>Cc: sip@ietf.org
> >>Subject: Re: [Sip] Status of SIP MIB
> >>
> >>bob,
> >>
> >>agreed.   and i think about the need to move the
> >>draft forward almost all the time.  the issue, for
> >>me personally, has been product development priorities. working on=20
> >>this mib is by no means my main job ;) i think i speak for my=20
> >>co-author as well.
> >>
> >>i alluded to this being the _2nd wglc_ for this draft.
> >>that was a result of us pausing, after the 1st wglc
> >>was concluded, and taking the extra step to migrate the
> >>mibs to be aligned with the 2nd rfc for sip (rfc3261).
> >>the release of rfc3261 came about in the same timeframe.
> >>so in that sense we are trying to provide the due diligence to make=20
> >>the mibs better.  an rfc would probably exist if we had not=20
> taken this=20
> >>step.  but it was the right thing to do.
> >>
> >>my comments about no ground swell relate to:
> >>1) the fact that we, over the life of this draft
> >>    [which is waaaayyyy too long], we've not gotten
> >>    any sense of urgency to get the mibs done.
> >>    i expect/hope this will change as sip becomes more and
> >>    more mainstream.   i may have been able to
> >>    justify more of my time for the draft work
> >>    had there been a more vocal response from the
> >>    sip community wrt the need for MIBs to be in
> >>    place sooner than later.
> >>
> >>2) our ability to solicit interest from the sip
> >>    community to review the various drafts has also
> >>    consistently been a problem.  to the point where
> >>    sometimes i felt we were offering candidate mib
> >>    objects in a vaccum.    having 2 last call reviews
> >>    has helped this issue though.    it is also a case
> >>    of most folks being very interested in the protocol
> >>    but not really being "MIB folk" ;)
> >>
> >>
> >>we will eventually get there with the draft (hopefully
> >>sooner than later ;).
> >>
> >>kevin
> >>
> >>Bob Natale wrote:
> >>
> >>>Hi Kevin,
> >>>
> >>>
> >>>
> >>>>my sense has been all along that there isn't a ground swell of
> >>>
> >>>immediate
> >>>
> >>>
> >>>>need for the SIP MIBs as of yet across the sip community.
> >>>
> >>>
> >>>Perhaps.  But good MIBs tend to beget good (SNMP[v3]-based)
> >>
> > management
> >=20
> >>>applications, not the other way around.  And the absence of good
> >>
> > MIBs
> >=20
> >>>for otherwise good technology (e.g., SIP) virtually=20
> guarantees that=20
> >>>other management solutions will be deployed, increasingly lowering
> >>
> > the
> >=20
> >>>sense of (but not the reality of) need for those good MIBs.
> >>>
> >>>Cheers,
> >>>BobN
> >>>
> >>>
> >>>_______________________________________________
> >>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>This list is for NEW development of the core SIP Protocol Use=20
> >>>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>>sipping@ietf.org for new developments on the application of sip
> >>>
> >>
> >=20
> >=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=20
> current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=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=20
> sip Use sipping@ietf.org for new developments on the=20
> application of sip
>=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



