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From: "Dale Worley" <dworley@nortel.com>
To: SIP <sip@ietf.org>
Organization: Nortel Networks
Date: Sun, 01 Feb 2009 14:53:56 -0500
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Subject: [Sip] Problems with History-Info
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To get discussion going for 2009, let me start with some complaints
about the History-Info header.

One problem is identifying which value in the History-Info header
corresponds to the request carrying the header, that is, which branch
of the History-Info request tree represents the current request.

Reading RFC 4244, it's clear that what is intended is that the
History-Info values will be written in depth-first tree-walk order,
and that syntactically *last* value corresponds to the request
carrying the History-Info header.  Neither of these principles is
stated explicitly as far as I can tell.

Difficulty arises when there is parallel forking, or any similar
situation when a request has multiple redirects or retargets that
become known in one processing step.  If one maintains the rule that
the enclosing request corresponds to the last History-Info value, one
must choose among:

(Assume that sip:1@example.com has been retargeted to
sip:1A@example.com and sip:1B@example.com.) 

1.each derived request carries only one of the new History-Info values

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1B@example.com>;index=1.1

2.each derived request carries only the new History-Info values that
  are "before" (in some ordering) the one for the current request

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1,
                      <sip:1B@example.com>;index=1.2

3.each derived request carries all of the new History-Info values, but
  in each request, they are ordered so that the value corresponding to
  the request is last

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1B@example.com>;index=1.1,
                      <sip:1A@example.com>;index=1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1,
                      <sip:1B@example.com>;index=1.2

Solutions 1 and 3 are troublesome because the generated History-Info
values are not consistent between the derived requests, and solutions
1 and 2 are troublesome because the downstream network elements are
given less information than is available about the forking of the
request.

Note that this isn't just a theoretical problem,
draft-rosenberg-sip-target-uri-delivery-00 is already running up
against it (section 5):  "Furthermore, if that URI had an 'index' of
N, the URIs with indices N.M for all M, are the registered contacts to
that URI."  If the I-D is taken at face value, solution 3 must be
used, giving different History-Info values on different derived
requests.

Let us then consider putting all known targets of a request in
History-Info, but not require that the last History-Info value
corresponds to the containing request.  In terms of the above example,
both derived requests would have the same History-Info header:

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1,
                      <sip:1B@example.com>;index=1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1,
                      <sip:1B@example.com>;index=1.2

If we do this, we need to be able to identify the value that
corresponds to the current request by some method other than the index
values and syntactic ordering.

In a perfect world, we could compare the request-URI with the URIs in
History-Info to see which one matches.  However, this fails if:

- two URIs in History-Info are the same (In theory, this should not
  happen, but it can happen sensibly if two derived requests differed
  substantially in some manner which could affect the ultimate
  resolution of the request-URI; e.g., caller preferences.)

- if an intermediate proxy has changed the request-URI but not updated
  History-Info (In this case, we want to know which History-Info value
  corresponds to the most recent parent request that was logged in
  History-Info.)

A more reliable solution would be to add a "current" marker to the
leaf of the History-Info tree that corresponds to the containing
request.  In the example above:

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1;current,
                      <sip:1B@example.com>;index=1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=1,
                      <sip:1A@example.com>;index=1.1,
                      <sip:1B@example.com>;index=1.2;current

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

A second problem with History-Info as it is currently structured is in
the handling of requests generated as a consequence of recursing on
3xx responses.

History-Info as it is now defined correctly shows the "forking
structure" of a request.  But we also want to know the "derivation" of
the current request-URI, the sequence of transformations from the
original request-URI (presumably the same as the To URI) to the
current request-URI.  Assuming that the transformations are done in
the ordinary manner by proxies, History-Info carries the needed
information.  From RFC 4244 section 4.5:

   UA1                      Proxy1                    Proxy2                       UA2

   |                          |                         |                           |
   |--INVITE sip:Bob@P1... -->|                         |                           |
      History-Info: <sip:Bob@P1.example.com>;index=1
   |                          |-INVITE sip:Bob@P2... -->|                           |
                                History-Info: <sip:Bob@P1.example.com>;index=1,
                                              <sip:Bob@P2.example.com>;index=1.1
   |                          |                         |                           |
   |                          |                         |-INVITE sip:User2@UA2...-->|
                                                          History-Info: <sip:Bob@P1.example.com>;index=1,
                                                                        <sip:Bob@P2.example.com>;index=1.1,
                                                                        <sip:User2@UA2.example.com>;index=1.1.1

In the last INVITE, starting with the last URI and working our way up
the tree, we see that the URI sequence is:

sip:Bob@P1.example.com -> sip:Bob@P2.example.com -> sip:User2@UA2.example.com

However, if one of the transformations is supplied by a 3xx response
(e.g., if the request is sent to a "redirect server"), the
History-Info data does not effectively show the source of the current
request-URI:

   UA1                      Proxy1                    Proxy2      UA2

   |                          |                         |          |
   |--INVITE sip:Bob@P1... -->|                         |          |
      History-Info: <sip:Bob@P1.example.com>;index=1
   |                          |-INVITE sip:Bob@P2... -->|          |
                                History-Info: <sip:Bob@P1.example.com>;index=1,
                                              <sip:Bob@P2.example.com>;index=1.1
   |                          |<-- 302
                                   Contact: <sip:User2@UA2.example.com>
   |                          |                         |          |
   |                          |-INVITE sip:User2@UA2...----------->|
                                History-Info: <sip:Bob@P1.example.com>;index=1,
                                              <sip:Bob@P2.example.com?Reason=SIP%3Bcause%3D302>;index=1.1,
                                              <sip:User2@UA2.example.com>;index=1.2

The apparent URI sequence is:

sip:Bob@P1.example.com -> sip:User2@UA2.example.com

There is no unambiguous way determine that the target
sip:User2@UA2.example.com came from the 302 response to the request to
sip:Bob@P2.example.com.

What is needed is a pointer from a History-Info value to the value
representing the request which yielded the 3xx response that cause the
value in question.  I suggest using a parameter named "3xx" for this
purpose, so that the last request in the above example becomes:

    INVITE sip:User2@UA2.example.com
    History-Info: <sip:Bob@P1.example.com>;index=1,
		  <sip:Bob@P2.example.com?Reason=SIP%3Bcause%3D302>;index=1.1,
		  <sip:User2@UA2.example.com>;index=1.2;3xx=1.1

Which this additional information present, the predecessor of a listed
URI is either the URI of its 3xx value (if present), or its parent in
the tree.  This allows us to recover the URI transformation sequence:

sip:Bob@P1.example.com -> sip:Bob@P2.example.com -> sip:User2@UA2.example.com

Dale


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Thread-Topic: Is Contact Header mandatory in REFER message?
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From: <sunil.bhagat@wipro.com>
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Subject: [Sip] Is Contact Header mandatory in REFER message?
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Hi,

=20

Is it mandatory to have a Contact Header field in REFER message?

=20

=20

RFC 3261            SIP: Session Initiation Protocol           June 2002

=20

=20

8.1.1.8 Contact
=20
   The Contact header field provides a SIP or SIPS URI that can be used
   to contact that specific instance of the UA for subsequent requests.
   The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog.  For the methods defined in this specification, that includes
   only the INVITE request.  For these requests, the scope of the
   Contact is global.  That is, the Contact header field value contains
   the URI at which the UA would like to receive requests, and this URI
   MUST be valid even if used in subsequent requests outside of any
   dialogs.
=20
   If the Request-URI or top Route header field value contains a SIPS
   URI, the Contact header field MUST contain a SIPS URI as well.

=20

=20

=20

=20

=20

RFC 3515                  The SIP Refer Method                April 2003
=20
=20
2. The REFER Method
=20
   REFER is a SIP method as defined by RFC 3261 [1].  The REFER method
   indicates that the recipient (identified by the Request-URI) should
   contact a third party using the contact information provided in the
   request.
=20
   Unless stated otherwise, the protocol for emitting and responding to
   a REFER request are identical to those for a BYE request in [1].  The
   behavior of SIP entities not implementing the REFER (or any other
   unknown) method is explicitly defined in [1].
=20
   A REFER request implicitly establishes a subscription to the refer
   event.  Event subscriptions are defined in [2].
=20
   A REFER request MAY be placed outside the scope of a dialog created
   with an INVITE.  REFER creates a dialog, and MAY be Record-Routed,
   hence MUST contain a single Contact header field value.  REFERs
   occurring inside an existing dialog MUST follow the Route/Record-
   Route logic of that dialog.
=20

=20

Regards,

Sunil

=20

=20


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Is it mandatory to have a Contact =
Header
field in REFER message?<o:p></o:p></span></font></p>

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>RFC =
3261&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
SIP: Session Initiation =
Protocol&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
June 2002<o:p></o:p></span></font></pre>

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>8.1.1.8 =
Contact<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; The Contact header field =
provides a SIP or SIPS URI that can be =
used<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; to contact that specific =
instance of the UA for subsequent =
requests.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; <b><span
style=3D'font-weight:bold'>The Contact header field MUST be present and =
contain exactly one =
SIP<o:p></o:p></span></b></span></font></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; or SIPS URI in =
any request that can result in the establishment of =
a<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; dialog.&nbsp; =
For the methods defined in this specification, that =
includes<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; only the INVITE =
request.</span></font></b>&nbsp; For these requests, the scope of =
the<o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Contact is global.&nbsp; That =
is, the Contact header field value =
contains<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; the URI at which the UA would =
like to receive requests, and this =
URI<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; MUST be valid even if used in =
subsequent requests outside of =
any<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
dialogs.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; If the Request-URI or top Route =
header field value contains a =
SIPS<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; URI, the Contact header field =
MUST contain a SIPS URI as well.<o:p></o:p></span></font></pre>

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

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

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

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>RFC =
3515&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The SIP Refer =
Method&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; April =
2003<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>2. The =
REFER Method<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; REFER is a SIP method as defined =
by RFC 3261 [1].&nbsp; The REFER =
method<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; indicates that the recipient =
(identified by the Request-URI) =
should<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; contact a third party using the =
contact information provided in =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
request.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Unless stated otherwise, the =
protocol for emitting and responding =
to<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; a REFER request are identical to =
those for a BYE request in [1].&nbsp; =
The<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; behavior of SIP entities not =
implementing the REFER (or any =
other<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; unknown) method is explicitly =
defined in [1].<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; A REFER request implicitly =
establishes a subscription to the =
refer<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; event.&nbsp; Event subscriptions =
are defined in [2].<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; <b><span
style=3D'font-weight:bold'>A REFER request MAY be placed outside the =
scope of a dialog =
created<o:p></o:p></span></b></span></font></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; with an =
INVITE.&nbsp; REFER creates a dialog, and MAY be =
Record-Routed,<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; hence MUST =
contain a single Contact header field value.</span></font></b>&nbsp; =
REFERs<o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; occurring inside an existing =
dialog MUST follow the =
Route/Record-<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Route logic of that =
dialog.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p=
>

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

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

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

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Thread-Topic: [Sip] Is Contact Header mandatory in REFER message?
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
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Subject: Re: [Sip] Is Contact Header mandatory in REFER message?
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=20
Yes


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of sunil.bhagat@wipro.com
	Sent: Monday, February 02, 2009 11:51 AM
	To: sip@ietf.org
	Subject: [Sip] Is Contact Header mandatory in REFER message?
=09
=09

	Hi,

	=20

	Is it mandatory to have a Contact Header field in REFER message?

	=20

	=20

	RFC 3261            SIP: Session Initiation Protocol
June 2002

	=20

	=20

	8.1.1.8 Contact
	=20
	   The Contact header field provides a SIP or SIPS URI that can
be used
	   to contact that specific instance of the UA for subsequent
requests.
	   The Contact header field MUST be present and contain exactly
one SIP
	   or SIPS URI in any request that can result in the
establishment of a
	   dialog.  For the methods defined in this specification, that
includes
	   only the INVITE request.  For these requests, the scope of
the
	   Contact is global.  That is, the Contact header field value
contains
	   the URI at which the UA would like to receive requests, and
this URI
	   MUST be valid even if used in subsequent requests outside of
any
	   dialogs.
	=20
	   If the Request-URI or top Route header field value contains a
SIPS
	   URI, the Contact header field MUST contain a SIPS URI as
well.

	=20

	=20

	=20

	=20

	=20

	RFC 3515                  The SIP Refer Method
April 2003
	=20
	=20
	2. The REFER Method
	=20
	   REFER is a SIP method as defined by RFC 3261 [1].  The REFER
method
	   indicates that the recipient (identified by the Request-URI)
should
	   contact a third party using the contact information provided
in the
	   request.
	=20
	   Unless stated otherwise, the protocol for emitting and
responding to
	   a REFER request are identical to those for a BYE request in
[1].  The
	   behavior of SIP entities not implementing the REFER (or any
other
	   unknown) method is explicitly defined in [1].
	=20
	   A REFER request implicitly establishes a subscription to the
refer
	   event.  Event subscriptions are defined in [2].
	=20
	   A REFER request MAY be placed outside the scope of a dialog
created
	   with an INVITE.  REFER creates a dialog, and MAY be
Record-Routed,
	   hence MUST contain a single Contact header field value.
REFERs
	   occurring inside an existing dialog MUST follow the
Route/Record-
	   Route logic of that dialog.
	=20

	=20

	Regards,

	Sunil

	=20

	=20


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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><SPAN =
class=3D077414708-02022009></SPAN><FONT=20
face=3DArial><FONT color=3D#0000ff><FONT size=3D2>Yes<SPAN=20
class=3D077414708-02022009></SPAN></FONT></FONT></FONT><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
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  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
  </B>sunil.bhagat@wipro.com<BR><B>Sent:</B> Monday, February 02, 2009 =
11:51=20
  AM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Is Contact =
Header=20
  mandatory in REFER message?<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Hi,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Is it =
mandatory to=20
  have a Contact Header field in REFER =
message?<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">RFC =
3261&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
SIP: Session Initiation =
Protocol&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
June 2002<o:p></o:p></SPAN></FONT></PRE>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">8.1.1.8 =
Contact<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; The Contact =
header field provides a SIP or SIPS URI that can be =
used<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; to contact that =
specific instance of the UA for subsequent =
requests.<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; <B><SPAN =
style=3D"FONT-WEIGHT: bold">The Contact header field MUST be present and =
contain exactly one =
SIP<o:p></o:p></SPAN></B></SPAN></FONT></PRE><PRE><B><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-WEIGHT: bold; =
FONT-SIZE: 10pt">&nbsp;&nbsp; or SIPS URI in any request that can result =
in the establishment of =
a<o:p></o:p></SPAN></FONT></B></PRE><PRE><B><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt">&nbsp;&nbsp; =
dialog.&nbsp; For the methods defined in this specification, that =
includes<o:p></o:p></SPAN></FONT></B></PRE><PRE><B><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-WEIGHT: bold; FONT-SIZE: =
10pt">&nbsp;&nbsp; only the INVITE request.</SPAN></FONT></B>&nbsp; For =
these requests, the scope of the<o:p></o:p></PRE><PRE><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp; Contact is global.&nbsp; That is, the Contact header =
field value contains<o:p></o:p></SPAN></FONT></PRE><PRE><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp; the URI at which the UA would like to receive =
requests, and this URI<o:p></o:p></SPAN></FONT></PRE><PRE><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp; MUST be valid even if used in subsequent requests =
outside of any<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; =
dialogs.<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; If the =
Request-URI or top Route header field value contains a =
SIPS<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; URI, the Contact =
header field MUST contain a SIPS URI as =
well.<o:p></o:p></SPAN></FONT></PRE>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">RFC =
3515&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The SIP Refer =
Method&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; April =
2003<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">2. The REFER =
Method<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; REFER is a =
SIP method as defined by RFC 3261 [1].&nbsp; The REFER =
method<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; indicates that the =
recipient (identified by the Request-URI) =
should<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; contact a third =
party using the contact information provided in =
the<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; =
request.<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; Unless stated =
otherwise, the protocol for emitting and responding =
to<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; a REFER request =
are identical to those for a BYE request in [1].&nbsp; =
The<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; behavior of SIP =
entities not implementing the REFER (or any =
other<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; unknown) method is =
explicitly defined in [1].<o:p></o:p></SPAN></FONT></PRE><PRE><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; A REFER =
request implicitly establishes a subscription to the =
refer<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; event.&nbsp; Event =
subscriptions are defined in =
[2].<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier New" =
size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; <B><SPAN =
style=3D"FONT-WEIGHT: bold">A REFER request MAY be placed outside the =
scope of a dialog =
created<o:p></o:p></SPAN></B></SPAN></FONT></PRE><PRE><B><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-WEIGHT: bold; =
FONT-SIZE: 10pt">&nbsp;&nbsp; with an INVITE.&nbsp; REFER creates a =
dialog, and MAY be =
Record-Routed,<o:p></o:p></SPAN></FONT></B></PRE><PRE><B><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-WEIGHT: bold; =
FONT-SIZE: 10pt">&nbsp;&nbsp; hence MUST contain a single Contact header =
field value.</SPAN></FONT></B>&nbsp; REFERs<o:p></o:p></PRE><PRE><FONT =
face=3D"Courier New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt">&nbsp;&nbsp; occurring inside an existing dialog MUST follow the =
Route/Record-<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: 10pt">&nbsp;&nbsp; Route logic =
of that dialog.<o:p></o:p></SPAN></FONT></PRE><PRE><FONT face=3D"Courier =
New" size=3D2><SPAN style=3D"FONT-SIZE: =
10pt"><o:p>&nbsp;</o:p></SPAN></FONT></PRE>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Regards,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Sunil<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTM=
L>

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Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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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
--===============1985489263==--

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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] [SIP] Max no of non-invite pending transaction within a dialog
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hanifa.mohammed wrote:
> Hi,
> 
> 1. Only one INVITE request may be pending within a dialog.  Else, 491 
> will be sent.
> 2. Only one non-INVITE request may be pending within a dialog (in one 
> direction): it is not possible to send UPDATE if no final response has 
> been received to PRACK.
> 
> Pl clarify whether the above points are right. Esp, the second point.

It is possible to have more than one non-invite transaction in progress, 
but it is usually unwise.

If they get out of order, so the 2nd arrives before the first, then the 
first will be rejected with a 500 response. There are a few cases where 
this is acceptable. But in most cases you would be well advised to 
restrict to one transaction at a time.

	Paul
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From sunil.bhagat@wipro.com  Wed Feb  4 01:39:55 2009
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Thread-Topic: [Sip] Is Contact Header mandatory in REFER message?
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Subject: Re: [Sip] Is Contact Header mandatory in REFER message?
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Thanks Sanjay.

=20

So, is the Contact Header mandatory for all SIP messages (Response and
Requests)?

=20

Regards,

Sunil

=20

________________________________

From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com]=20
Sent: Monday, February 02, 2009 2:18 PM
To: Sunil Bhagat (WT01 - Telecom Equipment); sip@ietf.org
Subject: RE: [Sip] Is Contact Header mandatory in REFER message?

=20

=20

Yes

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
sunil.bhagat@wipro.com
Sent: Monday, February 02, 2009 11:51 AM
To: sip@ietf.org
Subject: [Sip] Is Contact Header mandatory in REFER message?

Hi,

=20

Is it mandatory to have a Contact Header field in REFER message?

=20

=20

RFC 3261            SIP: Session Initiation Protocol           June 2002

=20

=20

8.1.1.8 Contact
=20
   The Contact header field provides a SIP or SIPS URI that can be used
   to contact that specific instance of the UA for subsequent requests.
   The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog.  For the methods defined in this specification, that includes
   only the INVITE request.  For these requests, the scope of the
   Contact is global.  That is, the Contact header field value contains
   the URI at which the UA would like to receive requests, and this URI
   MUST be valid even if used in subsequent requests outside of any
   dialogs.
=20
   If the Request-URI or top Route header field value contains a SIPS
   URI, the Contact header field MUST contain a SIPS URI as well.

=20

=20

=20

=20

=20

RFC 3515                  The SIP Refer Method                April 2003
=20
=20
2. The REFER Method
=20
   REFER is a SIP method as defined by RFC 3261 [1].  The REFER method
   indicates that the recipient (identified by the Request-URI) should
   contact a third party using the contact information provided in the
   request.
=20
   Unless stated otherwise, the protocol for emitting and responding to
   a REFER request are identical to those for a BYE request in [1].  The
   behavior of SIP entities not implementing the REFER (or any other
   unknown) method is explicitly defined in [1].
=20
   A REFER request implicitly establishes a subscription to the refer
   event.  Event subscriptions are defined in [2].
=20
   A REFER request MAY be placed outside the scope of a dialog created
   with an INVITE.  REFER creates a dialog, and MAY be Record-Routed,
   hence MUST contain a single Contact header field value.  REFERs
   occurring inside an existing dialog MUST follow the Route/Record-
   Route logic of that dialog.
=20

=20

Regards,

Sunil

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks =
Sanjay.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>So, is the Contact Header mandatory =
for
all SIP messages (Response and Requests)?<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p=
>

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

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

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
Sanjay Sinha
(sanjsinh) [mailto:sanjsinh@cisco.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, February =
02, 2009
2:18 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Sunil Bhagat (WT01 - =
Telecom
Equipment); <st1:PersonName =
w:st=3D"on">sip@ietf.org</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [Sip] Is =
Contact
Header mandatory in REFER message?</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

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

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b><span =
style=3D'font-weight:
bold'>On Behalf Of </span></b>sunil.bhagat@wipro.com<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, February =
02, 2009
11:51 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> <st1:PersonName =
w:st=3D"on">sip@ietf.org</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Is Contact =
Header
mandatory in REFER message?</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Is it mandatory to have a Contact =
Header
field in REFER message?<o:p></o:p></span></font></p>

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>RFC =
3261&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
SIP: Session Initiation =
Protocol&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
June 2002<o:p></o:p></span></font></pre>

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>8.1.1.8 =
Contact<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; The Contact header field =
provides a SIP or SIPS URI that can be =
used<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; to contact that specific =
instance of the UA for subsequent =
requests.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; <b><span
style=3D'font-weight:bold'>The Contact header field MUST be present and =
contain exactly one =
SIP<o:p></o:p></span></b></span></font></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; or SIPS URI in =
any request that can result in the establishment of =
a<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; dialog.&nbsp; =
For the methods defined in this specification, that =
includes<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; only the INVITE =
request.</span></font></b>&nbsp; For these requests, the scope of =
the<o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Contact is global.&nbsp; That =
is, the Contact header field value =
contains<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; the URI at which the UA would =
like to receive requests, and this =
URI<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; MUST be valid even if used in =
subsequent requests outside of =
any<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
dialogs.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; If the Request-URI or top Route =
header field value contains a =
SIPS<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; URI, the Contact header field =
MUST contain a SIPS URI as well.<o:p></o:p></span></font></pre>

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

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

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

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

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>RFC =
3515&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The SIP Refer =
Method&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; April =
2003<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>2. The =
REFER Method<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; REFER is a SIP method as defined =
by RFC 3261 [1].&nbsp; The REFER =
method<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; indicates that the recipient =
(identified by the Request-URI) =
should<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; contact a third party using the =
contact information provided in =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
request.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Unless stated otherwise, the =
protocol for emitting and responding =
to<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; a REFER request are identical to =
those for a BYE request in [1].&nbsp; =
The<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; behavior of SIP entities not =
implementing the REFER (or any =
other<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; unknown) method is explicitly =
defined in [1].<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; A REFER request implicitly =
establishes a subscription to the =
refer<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; event.&nbsp; Event subscriptions =
are defined in [2].<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; <b><span
style=3D'font-weight:bold'>A REFER request MAY be placed outside the =
scope of a dialog =
created<o:p></o:p></span></b></span></font></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; with an =
INVITE.&nbsp; REFER creates a dialog, and MAY be =
Record-Routed,<o:p></o:p></span></font></b></pre><pre><b><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-weight:bold'>&nbsp;&nbsp; hence MUST =
contain a single Contact header field value.</span></font></b>&nbsp; =
REFERs<o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; occurring inside an existing =
dialog MUST follow the =
Route/Record-<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; Route logic of that =
dialog.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p=
>

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

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

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

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Date: Wed, 04 Feb 2009 14:55:43 +0530
From: avinashkoneru <avinashkoneru@huawei.com>
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Subject: [Sip] How do we adress this SDP issue in SIP
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To SIP & SDP experts,

 

                               Iam having some doubts in RFC4566 (i.e
SDP)could you please explain them briefly

 

RFC 4566 PAGE 14

 

Hierarchical or layered encoding schemes are data streams where the encoding
from a single media source is split into a number of layers.

 

As per my understanding the above line says that  (Trade off between
bandwidth and quality )

Ex : I have a media file of 1 GB size .If I use layered encoding that 1 GB
media file could be made into 700k size with some loss of date  or 500k
size or 1GB only as per users choice? 

 

Such layered encodings are normally transmitted in multiple multicast groups
to allow multicast pruning

 

How would I know which layer is mapped to which multicast address (and size
of each layer)?..

 

I think this issue is addressed by the protocol above SDP (i.e in SIP or
SAP)?.. Could  anyone plz let me know .How and where this issue is addressed


 

 

 

Thanks and Regards,

Avinash Koneru

 

 

 

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>To SIP &amp; SDP =
experts,<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Iam having some doubts in RFC4566 (i.e SDP)could you please explain them
briefly<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>RFC 4566 PAGE 14<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'>Hierarchical or
layered encoding schemes are data streams where the encoding from a =
single
media source is split into a number of =
layers.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>As per =
my
understanding the above line says that&nbsp; (Trade off between =
bandwidth and
quality )<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>Ex : I =
have a
media file of 1 GB size .If I use layered encoding that 1 GB media file =
could
be made into 700k size with some loss of date &nbsp;or 500k &nbsp;size =
or 1GB
only as per users choice? <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>Such =
layered
encodings are normally transmitted in multiple multicast groups to allow =
multicast
pruning<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>How =
would I know
which layer is mapped to which multicast address (and size of each =
layer)?..<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>I think =
this
issue is addressed by the protocol above SDP (i.e in SIP or SAP)?.. =
Could&nbsp;
anyone plz let me know .How and where this issue is addressed =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:black'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>Thanks =
and
Regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>Avinash =
Koneru<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <avinashkoneru@huawei.com>, <sip@ietf.org>
References: <000501c986aa$8e3c5250$5811120a@china.huawei.com>
In-Reply-To: <000501c986aa$8e3c5250$5811120a@china.huawei.com>
Date: Wed, 4 Feb 2009 15:17:37 +0200
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Subject: Re: [Sip] How do we adress this SDP issue in SIP
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Hi,

You can start by looking at
http://tools.ietf.org/html/draft-ietf-mmusic-decoding-dependency-05

As for the layer description you can also look at the
draft-ietf-avt-rtp-svc.

 

Roni Even

 

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
avinashkoneru
Sent: Wednesday, February 04, 2009 11:26 AM
To: sip@ietf.org
Subject: [Sip] How do we adress this SDP issue in SIP

 

To SIP & SDP experts,

 

                               Iam having some doubts in RFC4566 (i.e
SDP)could you please explain them briefly

 

RFC 4566 PAGE 14

 

Hierarchical or layered encoding schemes are data streams where the encoding
from a single media source is split into a number of layers.

 

As per my understanding the above line says that  (Trade off between
bandwidth and quality )

Ex : I have a media file of 1 GB size .If I use layered encoding that 1 GB
media file could be made into 700k size with some loss of date  or 500k
size or 1GB only as per users choice? 

 

Such layered encodings are normally transmitted in multiple multicast groups
to allow multicast pruning

 

How would I know which layer is mapped to which multicast address (and size
of each layer)?..

 

I think this issue is addressed by the protocol above SDP (i.e in SIP or
SAP)?.. Could  anyone plz let me know .How and where this issue is addressed


 

 

 

Thanks and Regards,

Avinash Koneru

 

 

 

****************************************************************************
***********
This e-mail and attachments contain confidential information from HUAWEI,
which is intended only for the person or entity whose address is listed
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but not limited to, total or partial disclosure, reproduction, or
dissemination) by persons other than the intended recipient's) is
prohibited. If you receive this e-mail in error, please notify the sender by
phone or email immediately and delete it!

 


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<div class=3DSection1>

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

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>You can start by looking at <a
href=3D"http://tools.ietf.org/html/draft-ietf-mmusic-decoding-dependency-=
05">http://tools.ietf.org/html/draft-ietf-mmusic-decoding-dependency-05</=
a><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>As for the layer description you can also look at the
draft-ietf-avt-rtp-svc.<o:p></o:p></span></p>

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

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

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

<div>

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

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of =
</b>avinashkoneru<br>
<b>Sent:</b> Wednesday, February 04, 2009 11:26 AM<br>
<b>To:</b> sip@ietf.org<br>
<b>Subject:</b> [Sip] How do we adress this SDP issue in =
SIP<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>To
SIP &amp; SDP experts,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;
Iam having some doubts in RFC4566 (i.e SDP)could you please explain them =
briefly<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>RFC
4566 PAGE 14<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>Hierarchical or layered encoding schemes are data streams =
where
the encoding from a single media source is split into a number of =
layers.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>As per my understanding the above line says that&nbsp; =
(Trade off
between bandwidth and quality )<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>Ex : I have a media file of 1 GB size .If I use layered =
encoding
that 1 GB media file could be made into 700k size with some loss of date
&nbsp;or 500k &nbsp;size or 1GB only as per users choice? =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>Such layered encodings are normally transmitted in multiple
multicast groups to allow multicast pruning<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>How would I know which layer is mapped to which multicast =
address
(and size of each layer)?..<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>I think this issue is addressed by the protocol above SDP =
(i.e in
SIP or SAP)?.. Could&nbsp; anyone plz let me know .How and where this =
issue is
addressed <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>Thanks and Regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'>Avinash Koneru<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>*************=
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From john.elwell@siemens.com  Wed Feb  4 08:34:10 2009
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Date: Wed, 04 Feb 2009 16:31:25 +0000
From: "Elwell, John" <john.elwell@siemens.com>
To: sip@ietf.org
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Subject: [Sip] Question on loop detection
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Apologies if this has been discussed in the past.

Consider a domain proxy that is configured to parallel fork an INVITE
request to two targets. As a result it would forward the INVITE request
twice, and as far as I can see the two forwarded requests would in
general differ only in the following respects:
- different Request-URIs (the respective contact URIs);
- different top Via header field entries (different branch parameters);
- if applicable, different History-Info header field values.
=20
Supposing the two new targets are both reachable via the same edge
"proxy", which is actually implemented as a B2BUA (e.g., an SBC). The
edge B2BUA would receive one request and shortly afterwards would
receive the other request. The similarity and differences between the
two requests are such that, in accordance with RFC 3261, the second
request would be treated by the B2BUA (acting as a UAS) as a loop and be
rejected with 482, assuming it arrives within a given time period. For
TCP transport the second INVITE request would need to arrive before the
ACK relating to the first INVITE request, but for UDP transport the
window is extended by T4 seconds.

The text concerned in RFC 3261 is in 8.2.2.2:
"If the request has no tag in the To header field, the UAS core MUST
   check the request against ongoing transactions.  If the From tag,
   Call-ID, and CSeq exactly match those associated with an ongoing
   transaction, but the request does not match that transaction (based
   on the matching rules in Section 17.2.3), the UAS core SHOULD
   generate a 482 (Loop Detected) response and pass it to the server
   transaction."
in 17.2.3:
"The INVITE request matches a transaction if the Request-URI, To tag,
   From tag, Call-ID, CSeq, and top Via header field match those of the
   INVITE request which created the transaction.  In this case, the
   INVITE is a retransmission of the original one that created the
   transaction."
and in 17.1.2.2:
"Once the client transaction enters the "Completed" state, it MUST set
   Timer K to fire in T4 seconds for unreliable transports, and zero
   seconds for reliable transports.  The "Completed" state exists to
   buffer any additional response retransmissions that may be received
   (which is why the client transaction remains there only for
   unreliable transports).  T4 represents the amount of time the network
   will take to clear messages between client and server transactions.
   The default value of T4 is 5s."

Questions: Has this problem has been seen in practice? If so, what steps
have been taken to overcome it? If not, have I misinterpreted RFC 3261?

John

From theo@crazygreek.co.uk  Wed Feb  4 09:00:45 2009
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On Wed, Feb 4, 2009 at 4:31 PM, Elwell, John <john.elwell@siemens.com> wrote:

> Questions: Has this problem has been seen in practice? If so, what steps
> have been taken to overcome it? If not, have I misinterpreted RFC 3261?

The behaviour as you described it is correct when the SBC device is a
B2BUA with a single UA core for all sessions flowing through it, which
it isn't - or at least probably isn't what you want.

if instead you considered an SBC to have multiple logical UA cores,
one for each target that it's currently being an SBC for, then that
behaviour would be incorrect (as matching would be done by comparing
all other ongoing transactions within the same UAS core), and both
calls would flow through correctly.

this is how we have had our SBC behaving for the last 3 years without
problem [1].

 ~ Theo

1 - standard SBCs don't exist and it's all in our heads disclaimer applies :)

-- 
Theo Zourzouvillys
Chief Technical Officer
VoIP.co.uk - Commerce House, Telford Road, Bicester, OX26 4LD
Tel: +44 1908 764 196

From john.elwell@siemens.com  Wed Feb  4 09:24:58 2009
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Subject: Re: [Sip] Question on loop detection
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Theo,

Thanks. That is indeed one solution, but I am not sure all B2BUAs
deployable in such situations behave in this way, so I would welcome
other opinions.

John

> -----Original Message-----
> From: theo@crazygreek.co.uk [mailto:theo@crazygreek.co.uk] On=20
> Behalf Of Theo Zourzouvillys
> Sent: 04 February 2009 17:00
> To: Elwell, John
> Cc: sip@ietf.org; Armenio, Joao
> Subject: Re: [Sip] Question on loop detection
>=20
> On Wed, Feb 4, 2009 at 4:31 PM, Elwell, John=20
> <john.elwell@siemens.com> wrote:
>=20
> > Questions: Has this problem has been seen in practice? If=20
> so, what steps
> > have been taken to overcome it? If not, have I=20
> misinterpreted RFC 3261?
>=20
> The behaviour as you described it is correct when the SBC device is a
> B2BUA with a single UA core for all sessions flowing through it, which
> it isn't - or at least probably isn't what you want.
>=20
> if instead you considered an SBC to have multiple logical UA cores,
> one for each target that it's currently being an SBC for, then that
> behaviour would be incorrect (as matching would be done by comparing
> all other ongoing transactions within the same UAS core), and both
> calls would flow through correctly.
>=20
> this is how we have had our SBC behaving for the last 3 years without
> problem [1].
>=20
>  ~ Theo
>=20
> 1 - standard SBCs don't exist and it's all in our heads=20
> disclaimer applies :)
>=20
> --=20
> Theo Zourzouvillys
> Chief Technical Officer
> VoIP.co.uk - Commerce House, Telford Road, Bicester, OX26 4LD
> Tel: +44 1908 764 196
>=20

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Cc: sip@ietf.org, Theo Zourzouvillys <theo@voip.co.uk>, "Armenio, Joao" <joao.armenio@siemens.com>
Subject: Re: [Sip] Question on loop detection
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On Wed, 2009-02-04 at 17:24 +0000, Elwell, John wrote:
> Thanks. That is indeed one solution, but I am not sure all B2BUAs
> deployable in such situations behave in this way, so I would welcome
> other opinions.

I agree with Theo in results, though I'd phrase the answer differently
-- In this case, the SBC is acting as a pseudo-proxy, and so its loop
detection behavior must be different from what a true UAS would do; it
should be detecting loops in the same way that a proxy would.  Otherwise
you would get the functional problem that you describe.

Dale



From rjsparks@nostrum.com  Wed Feb  4 14:08:20 2009
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Cc: sip@ietf.org, "Armenio, Joao" <joao.armenio@siemens.com>
Subject: Re: [Sip] Question on loop detection
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You are describing a merge, not a loop, and as Dale pointed out, if  
the B2BUA is
playing the role of a proxy, it shouldn't merge detect. If it's being  
more than a proxy,
then it needs to choose one and reject the other with the same kind of  
local-policy
logic an actual endpoint would use.

RjS


On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:

> Apologies if this has been discussed in the past.
>
> Consider a domain proxy that is configured to parallel fork an INVITE
> request to two targets. As a result it would forward the INVITE  
> request
> twice, and as far as I can see the two forwarded requests would in
> general differ only in the following respects:
> - different Request-URIs (the respective contact URIs);
> - different top Via header field entries (different branch  
> parameters);
> - if applicable, different History-Info header field values.
>
> Supposing the two new targets are both reachable via the same edge
> "proxy", which is actually implemented as a B2BUA (e.g., an SBC). The
> edge B2BUA would receive one request and shortly afterwards would
> receive the other request. The similarity and differences between the
> two requests are such that, in accordance with RFC 3261, the second
> request would be treated by the B2BUA (acting as a UAS) as a loop  
> and be
> rejected with 482, assuming it arrives within a given time period. For
> TCP transport the second INVITE request would need to arrive before  
> the
> ACK relating to the first INVITE request, but for UDP transport the
> window is extended by T4 seconds.
>
> The text concerned in RFC 3261 is in 8.2.2.2:
> "If the request has no tag in the To header field, the UAS core MUST
>   check the request against ongoing transactions.  If the From tag,
>   Call-ID, and CSeq exactly match those associated with an ongoing
>   transaction, but the request does not match that transaction (based
>   on the matching rules in Section 17.2.3), the UAS core SHOULD
>   generate a 482 (Loop Detected) response and pass it to the server
>   transaction."
> in 17.2.3:
> "The INVITE request matches a transaction if the Request-URI, To tag,
>   From tag, Call-ID, CSeq, and top Via header field match those of the
>   INVITE request which created the transaction.  In this case, the
>   INVITE is a retransmission of the original one that created the
>   transaction."
> and in 17.1.2.2:
> "Once the client transaction enters the "Completed" state, it MUST set
>   Timer K to fire in T4 seconds for unreliable transports, and zero
>   seconds for reliable transports.  The "Completed" state exists to
>   buffer any additional response retransmissions that may be received
>   (which is why the client transaction remains there only for
>   unreliable transports).  T4 represents the amount of time the  
> network
>   will take to clear messages between client and server transactions.
>   The default value of T4 is 5s."
>
> Questions: Has this problem has been seen in practice? If so, what  
> steps
> have been taken to overcome it? If not, have I misinterpreted RFC  
> 3261?
>
> John
> _______________________________________________
> Sip mailing list  https://www.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 rfc-editor@rfc-editor.org  Wed Feb  4 15:34:33 2009
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Subject: [Sip] RFC 5411 on A Hitchhiker's Guide to the Session Initiation Protocol (SIP)
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5411

        Title:      A Hitchhiker's Guide to the 
                    Session Initiation Protocol (SIP) 
        Author:     J. Rosenberg
        Status:     Informational
        Date:       February 2009
        Mailbox:    jdrosen@cisco.com
        Pages:      39
        Characters: 96538
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-sip-hitchhikers-guide-06.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5411.txt

The Session Initiation Protocol (SIP) is the subject of numerous
specifications that have been produced by the IETF.  It can be
difficult to locate the right document, or even to determine the set
of Request for Comments (RFC) about SIP.  This specification serves
as a guide to the SIP RFC series.  It lists a current snapshot of the
specifications under the SIP umbrella, briefly summarizes each, and
groups them into categories.  This memo provides information for the 
Internet community.

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


INFORMATIONAL: This memo provides information for the Internet community.
It does not specify an Internet standard of any kind. Distribution of
this memo is unlimited.

This announcement is sent to the IETF-Announce and rfc-dist lists.
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The RFC Editor Team
USC/Information Sciences Institute



From abhatnagar192006@gmail.com  Wed Feb  4 19:44:06 2009
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>, "Armenio, Joao" <joao.armenio@siemens.com>
Subject: Re: [Sip] Question on loop detection
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I dont think the SBC will detect a loop. As per the RFC text above, a
loop situation will occur, only when the request does not match an
ongoing transaction at the UAS(as per the rules mentioned above), but
the call id, from tag and cseq match that of an ''ongoing''
transaction..
This situation will not arise in a parallel forking case, as each
forked branch creates a unique transaction of its own (and hence a
unique from tag).

i have a doubt on why a SBC should behave as a B2BUA? Should it not
behave as a proxy server between 2 sip domains?

aayush.

On 2/4/09, Robert Sparks <rjsparks@nostrum.com> wrote:
> You are describing a merge, not a loop, and as Dale pointed out, if
> the B2BUA is
> playing the role of a proxy, it shouldn't merge detect. If it's being
> more than a proxy,
> then it needs to choose one and reject the other with the same kind of
> local-policy
> logic an actual endpoint would use.
>
> RjS
>
>
> On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:
>
>> Apologies if this has been discussed in the past.
>>
>> Consider a domain proxy that is configured to parallel fork an INVITE
>> request to two targets. As a result it would forward the INVITE
>> request
>> twice, and as far as I can see the two forwarded requests would in
>> general differ only in the following respects:
>> - different Request-URIs (the respective contact URIs);
>> - different top Via header field entries (different branch
>> parameters);
>> - if applicable, different History-Info header field values.
>>
>> Supposing the two new targets are both reachable via the same edge
>> "proxy", which is actually implemented as a B2BUA (e.g., an SBC). The
>> edge B2BUA would receive one request and shortly afterwards would
>> receive the other request. The similarity and differences between the
>> two requests are such that, in accordance with RFC 3261, the second
>> request would be treated by the B2BUA (acting as a UAS) as a loop
>> and be
>> rejected with 482, assuming it arrives within a given time period. For
>> TCP transport the second INVITE request would need to arrive before
>> the
>> ACK relating to the first INVITE request, but for UDP transport the
>> window is extended by T4 seconds.
>>
>> The text concerned in RFC 3261 is in 8.2.2.2:
>> "If the request has no tag in the To header field, the UAS core MUST
>>   check the request against ongoing transactions.  If the From tag,
>>   Call-ID, and CSeq exactly match those associated with an ongoing
>>   transaction, but the request does not match that transaction (based
>>   on the matching rules in Section 17.2.3), the UAS core SHOULD
>>   generate a 482 (Loop Detected) response and pass it to the server
>>   transaction."
>> in 17.2.3:
>> "The INVITE request matches a transaction if the Request-URI, To tag,
>>   From tag, Call-ID, CSeq, and top Via header field match those of the
>>   INVITE request which created the transaction.  In this case, the
>>   INVITE is a retransmission of the original one that created the
>>   transaction."
>> and in 17.1.2.2:
>> "Once the client transaction enters the "Completed" state, it MUST set
>>   Timer K to fire in T4 seconds for unreliable transports, and zero
>>   seconds for reliable transports.  The "Completed" state exists to
>>   buffer any additional response retransmissions that may be received
>>   (which is why the client transaction remains there only for
>>   unreliable transports).  T4 represents the amount of time the
>> network
>>   will take to clear messages between client and server transactions.
>>   The default value of T4 is 5s."
>>
>> Questions: Has this problem has been seen in practice? If so, what
>> steps
>> have been taken to overcome it? If not, have I misinterpreted RFC
>> 3261?
>>
>> John
>> _______________________________________________
>> Sip mailing list  https://www.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://www.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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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : UA-Driven Privacy Mechanism for SIP
	Author(s)       : M. Munakata, et al.
	Filename        : draft-ietf-sip-ua-privacy-04.txt
	Pages           : 11
	Date            : 2009-02-04

This document defines a best current practice for a user agent to
generate an anonymous SIP message by utilizing mechanisms such as
GRUU (Globally Routable User Agent URIs) and TURN (Traversal Using
Relays around NAT) without the need for a privacy service defined in
RFC 3323.

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From john.elwell@siemens.com  Thu Feb  5 05:20:18 2009
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Date: Thu, 05 Feb 2009 13:19:54 +0000
From: "Elwell, John" <john.elwell@siemens.com>
To: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>, Shida Schubert <shida@ntt-at.com>, sip@ietf.org
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Mayumi,

I note that my previous comments of substance have all been taken into
account.

However, nits that I sent to you privately (see attached email) have not
been into account.

In addition I have detected a further nit:

In section 5.1.1 it says:
"...and refer to Section 4.2 for details on how to obtain
   an IP address through TURN."
Since IP addresses are not mentioned in the rest of 5.1.1, this seems
irrelevant. The words should be deleted.

When these nits have been fixed, in my opinion the document will be
ready to go.

John

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From: "Elwell, John" <john.elwell@siemens.com>
To: "Mayumi Munakata" <munakata.mayumi@lab.ntt.co.jp>,
	"Shida Schubert" <shida@ntt-at.com>

Mayumi, Shida,

I am sending you nits separately. My main comments have been posted to
the SIP list.

1. "by allowing user agent to
   take control of its privacy, rather than being completely dependent
   on external privacy service."
Change to
"by allowing a user agent to
             ^^
   take control of its privacy, rather than being completely dependent
   on an external privacy service."
      ^^^

2. "If revealing the domain name in Contact
   header field "
Change to:
"If revealing the domain name in the Contact
                                 ^^^^
   header field "

3. "A user agent that is not provided with a functional anonymous IP
   address through some administrative means, MUST obtain a relayed
   address if anonymity is desired (IP address of the media relay) for
   use in SDP and in Via header.  Such IP address is to be derived from
   a STUN relay server through TURN mechanism, which allows a STUN
   server to act as a media relay."
Change to:
"A user agent that is not provided with a functional anonymous IP
address through some administrative means MUST obtain a relayed address
(IP address of a relay) if anonymity is desired for use in SDP and in
the Via header field. Such an IP address is to be derived from a STUN
relay server through the TURN mechanism, which allows a STUN server to
act as a relay."
(Various changes, including deletion of "media", since the relay is also
used for Via (signalling)).

4. "A user agent SHOULD go with option 1 to conceal its domain name in
   From header field."
Insert "the" in front of "From".

5. "If user wants both anonymity and strong identity, use a third party
   anonymization service which issues AoR for the use in From address
   which also provides SIP-Identity."
Various changes:
"If the user wants both anonymity and strong identity, a solution would
be to use a third party anonymization service that issues an Address of
Record (AoR) for use in the From header field of a request and that also
provides a SIP-Identity Authentication Service."

6. "Without privacy considerations, the bottommost Via header field
added
   by a user agent"
Add "to a request".

7. "Via header SHOULD NOT include a host name."
Change to:
"The Via header field SHOULD NOT include a host name."

8. "Furthermore, if the provider of the caller intending to obscure its
   identity consists of a small number of people (e.g. small enterprise,
   SOHO), the domain name alone can reveal the identity of the caller
   when this specification is used."
Delete "when this specification is used". It is irrelevant whether this
specification is used - the domain name can in any case reveal
information.

9. "Same can be true when the provider is large, but the receiver of the
   call only knows few people from the source of call."
Change to:
"The same can be true when the provider is large but the receiver of the
 ^^^^^                                          ^ (delete commma)
   call only knows a few people from the source of call."
                   ^^

10. "The domain name in From header can be obscured as described in
   section 5.1.2, on contrary the Contact header needs to contain a
   valid domain name at all time to function properly."
Change to:
"The domain name in the From header field can be obscured as described
in
   section 5.1.2, whereas the Contact header field needs to contain a
   valid domain name at all times in order to function properly."

11. "If one wants to assure anonymization, it is recommended for the
user
   to seek and rely on third party anonymization service."
Change "third" to "a third".

12. "A third party anonymization service provides registrar and TURN
   service which has no affiliation with the caller's provider, allowing
   caller to completely withhold its identity."
Change to:
"A third party anonymization service provides registrar and TURN
   services that have no affiliation with the caller's provider,
allowing
   the caller to completely withhold its identity."

There are still several places where "header" is used instead of "header
field". A SIP message, like an HTTP message, has only one header yet
many header fields.

Regards,

John


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Date: Thu, 05 Feb 2009 13:19:54 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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To: aayush bhatnagar <abhatnagar192006@gmail.com>, Robert Sparks <rjsparks@nostrum.com>
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Subject: Re: [Sip] Question on loop detection
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Aayush,=20

> -----Original Message-----
> From: aayush bhatnagar [mailto:abhatnagar192006@gmail.com]=20
> Sent: 05 February 2009 03:44
> To: Robert Sparks
> Cc: Elwell, John; sip@ietf.org; Armenio, Joao
> Subject: Re: [Sip] Question on loop detection
>=20
> I dont think the SBC will detect a loop. As per the RFC text above, a
> loop situation will occur, only when the request does not match an
> ongoing transaction at the UAS(as per the rules mentioned above), but
> the call id, from tag and cseq match that of an ''ongoing''
> transaction..
> This situation will not arise in a parallel forking case, as each
> forked branch creates a unique transaction of its own (and hence a
> unique from tag).
[JRE] If an upstream proxy has forked, each branch will have the same
>From tag.

>=20
> i have a doubt on why a SBC should behave as a B2BUA? Should it not
> behave as a proxy server between 2 sip domains?
[JRE] I believe in practice SBCs do behave as B2BUAs, but perhaps not in
this respect. Behaving as proxies in this respect would indeed solve
this problem.

The reason for my original question is that I am aware of a device that
is a UAS and sends 482 when it ought not to, although according to RFC
3261 it is correct in doing so. By behaving as several UASs rather than
a single UAS, it would avoid the problem. But that made me question
whether the problem would also arise in other situations, e.g., with
B2BUAs such as SBCs. The answers I received to date suggest that proxy
behaviour in this respect would get round the problem, so perhaps that
is how such devices work in practice.

John


>=20
> aayush.
>=20
> On 2/4/09, Robert Sparks <rjsparks@nostrum.com> wrote:
> > You are describing a merge, not a loop, and as Dale pointed out, if
> > the B2BUA is
> > playing the role of a proxy, it shouldn't merge detect. If=20
> it's being
> > more than a proxy,
> > then it needs to choose one and reject the other with the=20
> same kind of
> > local-policy
> > logic an actual endpoint would use.
> >
> > RjS
> >
> >
> > On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:
> >
> >> Apologies if this has been discussed in the past.
> >>
> >> Consider a domain proxy that is configured to parallel=20
> fork an INVITE
> >> request to two targets. As a result it would forward the INVITE
> >> request
> >> twice, and as far as I can see the two forwarded requests would in
> >> general differ only in the following respects:
> >> - different Request-URIs (the respective contact URIs);
> >> - different top Via header field entries (different branch
> >> parameters);
> >> - if applicable, different History-Info header field values.
> >>
> >> Supposing the two new targets are both reachable via the same edge
> >> "proxy", which is actually implemented as a B2BUA (e.g.,=20
> an SBC). The
> >> edge B2BUA would receive one request and shortly afterwards would
> >> receive the other request. The similarity and differences=20
> between the
> >> two requests are such that, in accordance with RFC 3261, the second
> >> request would be treated by the B2BUA (acting as a UAS) as a loop
> >> and be
> >> rejected with 482, assuming it arrives within a given time=20
> period. For
> >> TCP transport the second INVITE request would need to arrive before
> >> the
> >> ACK relating to the first INVITE request, but for UDP transport the
> >> window is extended by T4 seconds.
> >>
> >> The text concerned in RFC 3261 is in 8.2.2.2:
> >> "If the request has no tag in the To header field, the UAS=20
> core MUST
> >>   check the request against ongoing transactions.  If the From tag,
> >>   Call-ID, and CSeq exactly match those associated with an ongoing
> >>   transaction, but the request does not match that=20
> transaction (based
> >>   on the matching rules in Section 17.2.3), the UAS core SHOULD
> >>   generate a 482 (Loop Detected) response and pass it to the server
> >>   transaction."
> >> in 17.2.3:
> >> "The INVITE request matches a transaction if the=20
> Request-URI, To tag,
> >>   From tag, Call-ID, CSeq, and top Via header field match=20
> those of the
> >>   INVITE request which created the transaction.  In this case, the
> >>   INVITE is a retransmission of the original one that created the
> >>   transaction."
> >> and in 17.1.2.2:
> >> "Once the client transaction enters the "Completed" state,=20
> it MUST set
> >>   Timer K to fire in T4 seconds for unreliable transports, and zero
> >>   seconds for reliable transports.  The "Completed" state exists to
> >>   buffer any additional response retransmissions that may=20
> be received
> >>   (which is why the client transaction remains there only for
> >>   unreliable transports).  T4 represents the amount of time the
> >> network
> >>   will take to clear messages between client and server=20
> transactions.
> >>   The default value of T4 is 5s."
> >>
> >> Questions: Has this problem has been seen in practice? If so, what
> >> steps
> >> have been taken to overcome it? If not, have I misinterpreted RFC
> >> 3261?
> >>
> >> John
> >> _______________________________________________
> >> Sip mailing list  https://www.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://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>=20

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Date: Thu, 05 Feb 2009 14:03:30 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Question on loop detection
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Robert,

It is true this is a merge rather than a loop situation, but the
conditions I describe are conditions for sending a 482 "loop detected".

John=20

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: 04 February 2009 22:08
> To: Elwell, John
> Cc: sip@ietf.org; Armenio, Joao
> Subject: Re: [Sip] Question on loop detection
>=20
> You are describing a merge, not a loop, and as Dale pointed out, if =20
> the B2BUA is
> playing the role of a proxy, it shouldn't merge detect. If=20
> it's being =20
> more than a proxy,
> then it needs to choose one and reject the other with the=20
> same kind of =20
> local-policy
> logic an actual endpoint would use.
>=20
> RjS
>=20
>=20
> On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:
>=20
> > Apologies if this has been discussed in the past.
> >
> > Consider a domain proxy that is configured to parallel fork=20
> an INVITE
> > request to two targets. As a result it would forward the INVITE =20
> > request
> > twice, and as far as I can see the two forwarded requests would in
> > general differ only in the following respects:
> > - different Request-URIs (the respective contact URIs);
> > - different top Via header field entries (different branch =20
> > parameters);
> > - if applicable, different History-Info header field values.
> >
> > Supposing the two new targets are both reachable via the same edge
> > "proxy", which is actually implemented as a B2BUA (e.g., an=20
> SBC). The
> > edge B2BUA would receive one request and shortly afterwards would
> > receive the other request. The similarity and differences=20
> between the
> > two requests are such that, in accordance with RFC 3261, the second
> > request would be treated by the B2BUA (acting as a UAS) as a loop =20
> > and be
> > rejected with 482, assuming it arrives within a given time=20
> period. For
> > TCP transport the second INVITE request would need to=20
> arrive before =20
> > the
> > ACK relating to the first INVITE request, but for UDP transport the
> > window is extended by T4 seconds.
> >
> > The text concerned in RFC 3261 is in 8.2.2.2:
> > "If the request has no tag in the To header field, the UAS core MUST
> >   check the request against ongoing transactions.  If the From tag,
> >   Call-ID, and CSeq exactly match those associated with an ongoing
> >   transaction, but the request does not match that=20
> transaction (based
> >   on the matching rules in Section 17.2.3), the UAS core SHOULD
> >   generate a 482 (Loop Detected) response and pass it to the server
> >   transaction."
> > in 17.2.3:
> > "The INVITE request matches a transaction if the=20
> Request-URI, To tag,
> >   From tag, Call-ID, CSeq, and top Via header field match=20
> those of the
> >   INVITE request which created the transaction.  In this case, the
> >   INVITE is a retransmission of the original one that created the
> >   transaction."
> > and in 17.1.2.2:
> > "Once the client transaction enters the "Completed" state,=20
> it MUST set
> >   Timer K to fire in T4 seconds for unreliable transports, and zero
> >   seconds for reliable transports.  The "Completed" state exists to
> >   buffer any additional response retransmissions that may=20
> be received
> >   (which is why the client transaction remains there only for
> >   unreliable transports).  T4 represents the amount of time the =20
> > network
> >   will take to clear messages between client and server=20
> transactions.
> >   The default value of T4 is 5s."
> >
> > Questions: Has this problem has been seen in practice? If so, what =20
> > steps
> > have been taken to overcome it? If not, have I misinterpreted RFC =20
> > 3261?
> >
> > John
> > _______________________________________________
> > Sip mailing list  https://www.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

From abhatnagar192006@gmail.com  Thu Feb  5 07:45:27 2009
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Date: Thu, 5 Feb 2009 21:15:04 +0530
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Cc: sip@ietf.org, "Armenio, Joao" <joao.armenio@siemens.com>
Subject: Re: [Sip] Question on loop detection
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>>[JRE] If an upstream proxy has forked, each branch will have the same
>>From tag.

True..agreed.

Another thing regarding section 17.2.3. The branch parameter will be
different for each forked branch, as specified by you inline. However,
the transaction matching rules of 17.2.3 you quote inline are
applicable only if the branch parameter is absent, or does not have a
magic cookie. If you have a magic cookie present, the matching wont
fail as per 17.2.3..and subsequently as per 8.2.2.2, there will be no
loop detection.

I suppose your implementation is not following the magic cookie paradigm?

>>[JRE] I believe in practice SBCs do behave as B2BUAs, but perhaps not in
>>this respect. Behaving as proxies in this respect would indeed solve
>>this problem.

Whether the SBC acts as a B2BUA or a proxy also depends on the
services that it is offering. If your SBC is handling media, QoS,
admission control,call screening and regulatory services such as
Lawful Intercept, then it will need to act as a B2BUA. It has no
choice.

However, if the SBC only needs to provide a point of interconnect
between trusted/non-trusted domains, then proxy behavior might be
sufficient.

On 2/5/09, Elwell, John <john.elwell@siemens.com> wrote:
> Robert,
>
> It is true this is a merge rather than a loop situation, but the
> conditions I describe are conditions for sending a 482 "loop detected".
>
> John
>
>> -----Original Message-----
>> From: Robert Sparks [mailto:rjsparks@nostrum.com]
>> Sent: 04 February 2009 22:08
>> To: Elwell, John
>> Cc: sip@ietf.org; Armenio, Joao
>> Subject: Re: [Sip] Question on loop detection
>>
>> You are describing a merge, not a loop, and as Dale pointed out, if
>> the B2BUA is
>> playing the role of a proxy, it shouldn't merge detect. If
>> it's being
>> more than a proxy,
>> then it needs to choose one and reject the other with the
>> same kind of
>> local-policy
>> logic an actual endpoint would use.
>>
>> RjS
>>
>>
>> On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:
>>
>> > Apologies if this has been discussed in the past.
>> >
>> > Consider a domain proxy that is configured to parallel fork
>> an INVITE
>> > request to two targets. As a result it would forward the INVITE
>> > request
>> > twice, and as far as I can see the two forwarded requests would in
>> > general differ only in the following respects:
>> > - different Request-URIs (the respective contact URIs);
>> > - different top Via header field entries (different branch
>> > parameters);
>> > - if applicable, different History-Info header field values.
>> >
>> > Supposing the two new targets are both reachable via the same edge
>> > "proxy", which is actually implemented as a B2BUA (e.g., an
>> SBC). The
>> > edge B2BUA would receive one request and shortly afterwards would
>> > receive the other request. The similarity and differences
>> between the
>> > two requests are such that, in accordance with RFC 3261, the second
>> > request would be treated by the B2BUA (acting as a UAS) as a loop
>> > and be
>> > rejected with 482, assuming it arrives within a given time
>> period. For
>> > TCP transport the second INVITE request would need to
>> arrive before
>> > the
>> > ACK relating to the first INVITE request, but for UDP transport the
>> > window is extended by T4 seconds.
>> >
>> > The text concerned in RFC 3261 is in 8.2.2.2:
>> > "If the request has no tag in the To header field, the UAS core MUST
>> >   check the request against ongoing transactions.  If the From tag,
>> >   Call-ID, and CSeq exactly match those associated with an ongoing
>> >   transaction, but the request does not match that
>> transaction (based
>> >   on the matching rules in Section 17.2.3), the UAS core SHOULD
>> >   generate a 482 (Loop Detected) response and pass it to the server
>> >   transaction."
>> > in 17.2.3:
>> > "The INVITE request matches a transaction if the
>> Request-URI, To tag,
>> >   From tag, Call-ID, CSeq, and top Via header field match
>> those of the
>> >   INVITE request which created the transaction.  In this case, the
>> >   INVITE is a retransmission of the original one that created the
>> >   transaction."
>> > and in 17.1.2.2:
>> > "Once the client transaction enters the "Completed" state,
>> it MUST set
>> >   Timer K to fire in T4 seconds for unreliable transports, and zero
>> >   seconds for reliable transports.  The "Completed" state exists to
>> >   buffer any additional response retransmissions that may
>> be received
>> >   (which is why the client transaction remains there only for
>> >   unreliable transports).  T4 represents the amount of time the
>> > network
>> >   will take to clear messages between client and server
>> transactions.
>> >   The default value of T4 is 5s."
>> >
>> > Questions: Has this problem has been seen in practice? If so, what
>> > steps
>> > have been taken to overcome it? If not, have I misinterpreted RFC
>> > 3261?
>> >
>> > John
>> > _______________________________________________
>> > Sip mailing list  https://www.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://www.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 john.elwell@siemens.com  Thu Feb  5 08:48:11 2009
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Date: Thu, 05 Feb 2009 16:47:46 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Question on loop detection
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Aayush,=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of aayush bhatnagar
> Sent: 05 February 2009 15:45
> To: Elwell, John
> Cc: sip@ietf.org; Armenio, Joao
> Subject: Re: [Sip] Question on loop detection
>=20
> >>[JRE] If an upstream proxy has forked, each branch will=20
> have the same
> >>From tag.
>=20
> True..agreed.
>=20
> Another thing regarding section 17.2.3. The branch parameter will be
> different for each forked branch, as specified by you inline. However,
> the transaction matching rules of 17.2.3 you quote inline are
> applicable only if the branch parameter is absent, or does not have a
> magic cookie. If you have a magic cookie present, the matching wont
> fail as per 17.2.3..and subsequently as per 8.2.2.2, there will be no
> loop detection.
[JRE] Whether or not the branch parameter matches, the Request-URI will
not match. So therefore the transactions do not match and the rules for
182 come into play.

>=20
> I suppose your implementation is not following the magic=20
> cookie paradigm?
>=20
> >>[JRE] I believe in practice SBCs do behave as B2BUAs, but=20
> perhaps not in
> >>this respect. Behaving as proxies in this respect would indeed solve
> >>this problem.
>=20
> Whether the SBC acts as a B2BUA or a proxy also depends on the
> services that it is offering. If your SBC is handling media, QoS,
> admission control,call screening and regulatory services such as
> Lawful Intercept, then it will need to act as a B2BUA. It has no
> choice.
[JRE] I agree, but it might behave as a proxy with respect to this
particular issue.

John

>=20
> However, if the SBC only needs to provide a point of interconnect
> between trusted/non-trusted domains, then proxy behavior might be
> sufficient.
>=20
> On 2/5/09, Elwell, John <john.elwell@siemens.com> wrote:
> > Robert,
> >
> > It is true this is a merge rather than a loop situation, but the
> > conditions I describe are conditions for sending a 482=20
> "loop detected".
> >
> > John
> >
> >> -----Original Message-----
> >> From: Robert Sparks [mailto:rjsparks@nostrum.com]
> >> Sent: 04 February 2009 22:08
> >> To: Elwell, John
> >> Cc: sip@ietf.org; Armenio, Joao
> >> Subject: Re: [Sip] Question on loop detection
> >>
> >> You are describing a merge, not a loop, and as Dale pointed out, if
> >> the B2BUA is
> >> playing the role of a proxy, it shouldn't merge detect. If
> >> it's being
> >> more than a proxy,
> >> then it needs to choose one and reject the other with the
> >> same kind of
> >> local-policy
> >> logic an actual endpoint would use.
> >>
> >> RjS
> >>
> >>
> >> On Feb 4, 2009, at 10:31 AM, Elwell, John wrote:
> >>
> >> > Apologies if this has been discussed in the past.
> >> >
> >> > Consider a domain proxy that is configured to parallel fork
> >> an INVITE
> >> > request to two targets. As a result it would forward the INVITE
> >> > request
> >> > twice, and as far as I can see the two forwarded=20
> requests would in
> >> > general differ only in the following respects:
> >> > - different Request-URIs (the respective contact URIs);
> >> > - different top Via header field entries (different branch
> >> > parameters);
> >> > - if applicable, different History-Info header field values.
> >> >
> >> > Supposing the two new targets are both reachable via the=20
> same edge
> >> > "proxy", which is actually implemented as a B2BUA (e.g., an
> >> SBC). The
> >> > edge B2BUA would receive one request and shortly afterwards would
> >> > receive the other request. The similarity and differences
> >> between the
> >> > two requests are such that, in accordance with RFC 3261,=20
> the second
> >> > request would be treated by the B2BUA (acting as a UAS) as a loop
> >> > and be
> >> > rejected with 482, assuming it arrives within a given time
> >> period. For
> >> > TCP transport the second INVITE request would need to
> >> arrive before
> >> > the
> >> > ACK relating to the first INVITE request, but for UDP=20
> transport the
> >> > window is extended by T4 seconds.
> >> >
> >> > The text concerned in RFC 3261 is in 8.2.2.2:
> >> > "If the request has no tag in the To header field, the=20
> UAS core MUST
> >> >   check the request against ongoing transactions.  If=20
> the From tag,
> >> >   Call-ID, and CSeq exactly match those associated with=20
> an ongoing
> >> >   transaction, but the request does not match that
> >> transaction (based
> >> >   on the matching rules in Section 17.2.3), the UAS core SHOULD
> >> >   generate a 482 (Loop Detected) response and pass it to=20
> the server
> >> >   transaction."
> >> > in 17.2.3:
> >> > "The INVITE request matches a transaction if the
> >> Request-URI, To tag,
> >> >   From tag, Call-ID, CSeq, and top Via header field match
> >> those of the
> >> >   INVITE request which created the transaction.  In this=20
> case, the
> >> >   INVITE is a retransmission of the original one that created the
> >> >   transaction."
> >> > and in 17.1.2.2:
> >> > "Once the client transaction enters the "Completed" state,
> >> it MUST set
> >> >   Timer K to fire in T4 seconds for unreliable=20
> transports, and zero
> >> >   seconds for reliable transports.  The "Completed"=20
> state exists to
> >> >   buffer any additional response retransmissions that may
> >> be received
> >> >   (which is why the client transaction remains there only for
> >> >   unreliable transports).  T4 represents the amount of time the
> >> > network
> >> >   will take to clear messages between client and server
> >> transactions.
> >> >   The default value of T4 is 5s."
> >> >
> >> > Questions: Has this problem has been seen in practice?=20
> If so, what
> >> > steps
> >> > have been taken to overcome it? If not, have I misinterpreted RFC
> >> > 3261?
> >> >
> >> > John
> >> > _______________________________________________
> >> > Sip mailing list  https://www.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=20
> application of sip
> >>
> >>
> > _______________________________________________
> > Sip mailing list  https://www.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://www.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

From dworley@nortel.com  Thu Feb  5 12:54:49 2009
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From: "Dale Worley" <dworley@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Question on loop detection
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On Thu, 2009-02-05 at 14:03 +0000, Elwell, John wrote:
> It is true this is a merge rather than a loop situation, but the
> conditions I describe are conditions for sending a 482 "loop detected".

RFC 3261 is correct but not at all clear in this regard.  There are
*two* error conditions, and RFC 3261 calls both "482 (Loop Detected)".
But in reality, there is "482 (Loop Detected)" and "482 (Merged
Request)".

482 Merged Request is generated by UASs, as described in section
8.2.2.2, and it is based on noticing that two incoming requests
originated as the same UAC request.  But the gotcha is that 8.2.2.2
describes UAS behavior, not proxy behavior.

482 Loop Detected is generated by proxies, as described in section 16.3
item 4, and is based on detecting that this request has an "upstream
ancestor" request that passed through this same proxy.

In regard to a B2BUA acting as an SBC, or a "quasi-proxy" as I like to
call it, it needs to apply the loop detection logic, not the merged
request logic.

Dale



From eburger@standardstrack.com  Fri Feb  6 10:49:06 2009
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Cc: SIP List <SIP@ietf.org>
Subject: [Sip] Comments on draft-ietf-sip-body-handling-05
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Good work and almost done!

General
=======
Many of the rules for multipart/mumble say things like "all parts must  
be foo." This construction is not strictly correct. For example, in  
multipart/alternative, I could have a multipart/mixed. Saying  
something like "all parts in a multipart/alternative must be marked  
foo" means that ALL parts, including walking down the multipart/mixed  
contained therein, must be marked foo.  What we really want to say is  
"all top-level parts in a multipart/alternative must be marked foo."  
That means, for example, if one of the parts is a multipart/mixed, the  
multipart/mixed needs to have the foo marking, but things within the  
multipart/mixed, I presume, will follow the appropriate rules for  
multipart/mixed and the body's context.


Section 3.2
===========
Please make clear that binary *SIP* message body parts MUST be binary,  
as the text currently says. However, it is quite possible, and should  
be legal, to encode application parts transported in SIP, such as INFO  
or NOTIFY bodies.  For example, one might forward a base64 e-mail or a  
uuencoded file.  Since that would be an application-level message,  
there is no reason to burden the application with decoding the body  
part to get it to binary, just for the other side to re-encode it.


Section 6.2
===========
What confused me is section 6.2 starts with what looks like a  
normative statement, but it really is a forward reference. To me,  
there is no reference to which 'same' we are referring to.  The text  
then seems to jump to what looks like a special case for 'session' and  
'early-session'.  I would suggest the following:


6.2.  UA Behavior to Generate 'multipart/alternative' Message Bodies

   Section 8.2 mandates all top-level body parts within a 'multipart/ 
alternative' have the same disposition type as the enclosing  
'multipart/alternative'.

    The 'session' and 'early-session' [RFC3959] disposition types  
require that all body parts of a 'multipart/alternative' body have  
different content types.   Consequently, for the 'session' and 'early- 
session' disposition  types, each top-level body type within the  
'multipart/alternative' body MUST be different.  Otherwise, the rules  
for the specific disposition type apply.
       NOTE: One could envision situations where a UA presents  
multiple alternatives with the same body type, where the receiving UA  
picks the best or richest alternative it can use.


Section 6.3
===========
Ah, but there ARE handling rules for multipart/alternative.  I would  
replace this section with:

    When a UA receives a message with a 'multipart/alternative' body  
part, if the body part has a 'required' handling, then the UA MUST  
pick the best alternative body part to use.  If the UA is unable to  
use any of the body parts, the UA MUST reject the request with a 415  
error response.  This could occur if the UA did not understand any of  
the top-level body parts within the 'multipart/alternative'.

    The handling marking for top-level body parts in a 'multipart/ 
alternative' has no meaning.  Therefore, the UA MUST ignore any  
'handling' tags in the top-level body parts of the 'multipart/ 
alternative' body part.  Once the UA picks one of the top-level body  
parts to use, then the UA MUST use the appropriate processing rules  
for the 'handling' tags within the selected top-level body part.


Section 8.2
===========
For multipart/alternative, a top-level body type with  
'handling=required' makes no sense.  By definition, using multipart/ 
alternative means we expect the UA to not be able to handle one or  
more of the body parts - we just do not know which one.  I would  
remove the note and incorporate the gist into the text:

    If the sending UA requires the handling of a 'multipart/ 
alternative' body as a whole, the UA MUST set the 'handling' parameter  
of the 'multipart/alternative' body to 'required'.  Otherwise, the UA  
MUST set the 'handling' parameter to 'optional'.  The sending UA MUST  
set all the top-level body parts within the 'multipart/alternative' to  
'handling=optional'.


Section 8.2
===========
For multipart/related, I disagree that if any part is required, the / 
related is required.  The last sentence is a non-sequitur.  Either I  
mandate doing something with the 'multipart/related' body or I do  
not.  I could see, for example, saying that processing the multipart/ 
related is optional, but if you do process it, one or more of the sub- 
parts is required. In this case, the multipart/related is optional,  
yet one of the sub-parts is required.

How about:

    If the sending UA requires the handling of a 'multipart/related'  
body as a whole, the UA MUST set the 'handling' parameter of the  
'multipart/related' body to 'required'.  Otherwise, the UA MUST set  
the handling to 'optional'.  The UA sets the 'handling' parameters of  
the body parts within the 'multipart/related' body independently from  
the 'handling' parameter o the 'multipart/related' body.  If the  
sending UA requires the handling of a particular body, the UA MUST set  
the 'handling' parameter of that body part 'required'.  Otherwise, the  
UA MUST set the parameter to 'optional'.



Section 8.2
===========
General: I really do not like the construction "If the UA requires the  
handling of foo the UA MUST mark it with 'handling=required'."  The  
reason I do not like this is this is precisely what RFC 3459 says to  
do.  All this draft does is repeat what is in that draft.  Saying "You  
MUST do what you MUST do" seems redundant and error prone.



Section 8.3
===========
This section is redundant with section 6.3.  Remove section 8.3, as  
section 6.3 talks about processing.


Section 8.4
===========
OK, I'll bite: what if I do not feel like returning a 415?  Either  
enumerate the situation where I MUST return a 415 or enumerate the  
situations where I would return something other than 415.



Section 9.2
===========
Nit in last line of Note: s/constrain/constraint/


Section 9.4
===========
Since RFC 3459 updated 3204, would not the normative current reference  
be to 3459?


Section 11
==========
I would offer we point out the potential for DoS attacks from complex  
MIME bodies.  UA's need to be aware of potential stack depth overflow,  
line length overflow, pointers to missing content, incorrect content- 
lengths, correct content-lengths that span what a parser that does not  
look at content-lengths might consider to be content boundaries with  
other body parts snuck in.
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From: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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Subject: Re: [Sip] Nits in draft-ietf-sip-ua-privacy-04
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John,

Thank you for the reminder.
I am so sorry that I totally missed those
nits comments.

I will correct the draft to incorporate
your comments and submit 05 version soon.

Mayumi


> Mayumi,
> 
> I note that my previous comments of substance have all been taken into
> account.
> 
> However, nits that I sent to you privately (see attached email) have not
> been into account.
> 
> In addition I have detected a further nit:
> 
> In section 5.1.1 it says:
> "...and refer to Section 4.2 for details on how to obtain
>    an IP address through TURN."
> Since IP addresses are not mentioned in the rest of 5.1.1, this seems
> irrelevant. The words should be deleted.
> 
> When these nits have been fixed, in my opinion the document will be
> ready to go.
> 
> John
> 
> 
> ------------------------------------------------------------------------
> 
> Subject:
> Nits in draft-ietf-sip-ua-privacy-03
> From:
> "Elwell, John" <john.elwell@siemens.com>
> Date:
> Tue, 11 Nov 2008 14:35:25 -0000
> To:
> "Mayumi Munakata" <munakata.mayumi@lab.ntt.co.jp>, "Shida Schubert" 
> <shida@ntt-at.com>
> 
> To:
> "Mayumi Munakata" <munakata.mayumi@lab.ntt.co.jp>, "Shida Schubert" 
> <shida@ntt-at.com>
> 
> 
> Mayumi, Shida,
> 
> I am sending you nits separately. My main comments have been posted to
> the SIP list.
> 
> 1. "by allowing user agent to
>    take control of its privacy, rather than being completely dependent
>    on external privacy service."
> Change to
> "by allowing a user agent to
>              ^^
>    take control of its privacy, rather than being completely dependent
>    on an external privacy service."
>       ^^^
> 
> 2. "If revealing the domain name in Contact
>    header field "
> Change to:
> "If revealing the domain name in the Contact
>                                  ^^^^
>    header field "
> 
> 3. "A user agent that is not provided with a functional anonymous IP
>    address through some administrative means, MUST obtain a relayed
>    address if anonymity is desired (IP address of the media relay) for
>    use in SDP and in Via header.  Such IP address is to be derived from
>    a STUN relay server through TURN mechanism, which allows a STUN
>    server to act as a media relay."
> Change to:
> "A user agent that is not provided with a functional anonymous IP
> address through some administrative means MUST obtain a relayed address
> (IP address of a relay) if anonymity is desired for use in SDP and in
> the Via header field. Such an IP address is to be derived from a STUN
> relay server through the TURN mechanism, which allows a STUN server to
> act as a relay."
> (Various changes, including deletion of "media", since the relay is also
> used for Via (signalling)).
> 
> 4. "A user agent SHOULD go with option 1 to conceal its domain name in
>    From header field."
> Insert "the" in front of "From".
> 
> 5. "If user wants both anonymity and strong identity, use a third party
>    anonymization service which issues AoR for the use in From address
>    which also provides SIP-Identity."
> Various changes:
> "If the user wants both anonymity and strong identity, a solution would
> be to use a third party anonymization service that issues an Address of
> Record (AoR) for use in the From header field of a request and that also
> provides a SIP-Identity Authentication Service."
> 
> 6. "Without privacy considerations, the bottommost Via header field
> added
>    by a user agent"
> Add "to a request".
> 
> 7. "Via header SHOULD NOT include a host name."
> Change to:
> "The Via header field SHOULD NOT include a host name."
> 
> 8. "Furthermore, if the provider of the caller intending to obscure its
>    identity consists of a small number of people (e.g. small enterprise,
>    SOHO), the domain name alone can reveal the identity of the caller
>    when this specification is used."
> Delete "when this specification is used". It is irrelevant whether this
> specification is used - the domain name can in any case reveal
> information.
> 
> 9. "Same can be true when the provider is large, but the receiver of the
>    call only knows few people from the source of call."
> Change to:
> "The same can be true when the provider is large but the receiver of the
>  ^^^^^                                          ^ (delete commma)
>    call only knows a few people from the source of call."
>                    ^^
> 
> 10. "The domain name in From header can be obscured as described in
>    section 5.1.2, on contrary the Contact header needs to contain a
>    valid domain name at all time to function properly."
> Change to:
> "The domain name in the From header field can be obscured as described
> in
>    section 5.1.2, whereas the Contact header field needs to contain a
>    valid domain name at all times in order to function properly."
> 
> 11. "If one wants to assure anonymization, it is recommended for the
> user
>    to seek and rely on third party anonymization service."
> Change "third" to "a third".
> 
> 12. "A third party anonymization service provides registrar and TURN
>    service which has no affiliation with the caller's provider, allowing
>    caller to completely withhold its identity."
> Change to:
> "A third party anonymization service provides registrar and TURN
>    services that have no affiliation with the caller's provider,
> allowing
>    the caller to completely withhold its identity."
> 
> There are still several places where "header" is used instead of "header
> field". A SIP message, like an HTTP message, has only one header yet
> many header fields.
> 
> Regards,
> 
> John
> 


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Hi All.

=20

I have a small query regarding the Contact Header while building a REFER
message.

RFC 3515 specifies that Contact Header is mandatory in REFER message.=20

If Contact Header is missing in REFER, can we populate its value from
the "From" Header?

=20

If yes, Can the same be done for SUBSCRIBE and NOTIFY messages?

=20

Thanks,

Sunil

=20


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi =
All.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I have a small query regarding the =
Contact
Header while building a REFER message.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>RFC 3515 specifies that Contact =
Header is
mandatory in REFER message. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If Contact Header is missing in =
REFER, can
we populate its value from the &#8220;From&#8221; =
Header?<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If yes, Can the same be done for =
SUBSCRIBE
and NOTIFY messages?<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks,<o:p></o:p></span></font></p>=


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

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

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Subject: Re: [Sip] Can we use From Header's value to build Contact in REFER
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Generally speaking, no.

The =46rom header field is likely (but not guaranteed) to contain =20
something with AOR properties, and you want an identifier that =20
corresponds to
exactly the endpoint that sent the REFER.

You're better off rejecting such a REFER as malformed.

The same is true of SUBSCRIBE, and because of forking, initial NOTIFYs.

You could consider being more lenient if Contact is missing in =20
subsequent NOTIFYs, using the remote target already stored in the =20
dialog, but
remember that you'd be writing code to deal with a peer that is not =20
following the protocol and that other exceptions are quite likely.

RjS

On Feb 8, 2009, at 11:17 PM, <sunil.bhagat@wipro.com> =
<sunil.bhagat@wipro.com=20
 > wrote:

> Hi All.
>
> I have a small query regarding the Contact Header while building a =20
> REFER message.
> RFC 3515 specifies that Contact Header is mandatory in REFER message.
> If Contact Header is missing in REFER, can we populate its value =20
> from the =93From=94 Header?
>
> If yes, Can the same be done for SUBSCRIBE and NOTIFY messages?
>
> Thanks,
> Sunil
>
> _______________________________________________
> Sip mailing list  https://www.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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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Generally speaking, =
no.<div><br><div>The =46rom header field is likely (but not guaranteed) =
to contain something with AOR properties, and you want an identifier =
that corresponds to&nbsp;</div><div>exactly the endpoint that sent the =
REFER.</div><div><br></div><div>You're better off rejecting such a REFER =
as malformed.</div><div><br></div><div>The same is true of SUBSCRIBE, =
and because of forking, initial NOTIFYs.</div><div><br></div><div>You =
could consider being more lenient if Contact is missing in subsequent =
NOTIFYs, using the remote target already stored in the dialog, =
but</div><div>remember that you'd be writing code to deal with a peer =
that is not following the protocol and that other exceptions are quite =
likely.</div><div><br></div><div>RjS</div><div><br><div><div><div>On Feb =
8, 2009, at 11:17 PM, &lt;<a =
href=3D"mailto:sunil.bhagat@wipro.com">sunil.bhagat@wipro.com</a>> =
&lt;<a href=3D"mailto:sunil.bhagat@wipro.com">sunil.bhagat@wipro.com</a>> =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
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-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple"><div class=3D"Section1"><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; ">Hi All.<o:p></o:p></span></font></div><div =
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Roman'; "><font size=3D"2" color=3D"navy" face=3D"Arial"><span =
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"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; ">I have a small query regarding the Contact Header while =
building a REFER message.<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman'; "><font size=3D"2" color=3D"navy" face=3D"Arial"><span =
style=3D"font-size: 10pt; font-family: Arial; color: navy; ">RFC 3515 =
specifies that Contact Header is mandatory in REFER =
message.<o:p></o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; ">If Contact Header is missing in REFER, can we populate =
its value from the =93From=94 Header?<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman'; "><font size=3D"2" color=3D"navy" face=3D"Arial"><span =
style=3D"font-size: 10pt; font-family: Arial; color: navy; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; ">If yes, Can the same be done for SUBSCRIBE and NOTIFY =
messages?<o:p></o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; "><o:p>&nbsp;</o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman'; "><font size=3D"2" color=3D"navy" face=3D"Arial"><span =
style=3D"font-size: 10pt; font-family: Arial; color: navy; =
">Thanks,<o:p></o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"navy" =
face=3D"Arial"><span style=3D"font-size: 10pt; font-family: Arial; =
color: navy; ">Sunil<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman'; "><font size=3D"2" face=3D"Arial"><span style=3D"font-size: =
10pt; font-family: Arial; =
"><o:p>&nbsp;</o:p></span></font></div></div>_____________________________=
__________________<br>Sip mailing list &nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/sip" style=3D"color: blue; =
text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW =
development of the core SIP Protocol<br>Use<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:sip-implementors@cs.columbia.edu" style=3D"color: blue; =
text-decoration: underline; ">sip-implementors@cs.columbia.edu</a><span =
class=3D"Apple-converted-space">&nbsp;</span>for questions on current =
sip<br>Use<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:sipping@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">sipping@ietf.org</a><span =
class=3D"Apple-converted-space">&nbsp;</span>for new developments on the =
application of =
sip</div></span></blockquote></div><br></div></div></div></body></html>=

--Apple-Mail-62-335886395--

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Subject: [Sip] [Technical Errata Reported] RFC4028 (1681)
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The following errata report has been submitted for RFC4028,
"Session Timers in the Session Initiation Protocol (SIP)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=4028&eid=1681

--------------------------------------
Type: Technical
Reported by: Muthu Arul Mozhi <mperumal@cisco.com>

Section: 13

Original Text
-------------
In section 13 (Example Call Flow) the From tag never changes 
between the initial INVITE message and the subsequent INVITE 
messages sent after receiving a 422:

message 1
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
   Supported: timer
   Session-Expires: 50
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp
   Content-Length: 142

message 4
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
   Supported: timer
   Session-Expires: 3600
   Min-SE: 3600
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314160 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp
   Content-Length: 142

message 10
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
   Supported: timer
   Session-Expires: 4000
   Min-SE: 4000
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314161 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp

However, as per RFC 3261, if an initial INVITE generates a non-2xx final
response, that terminates all sessions and all dialogs that were created. 

Hence, these are not re-INVITE messages, rather new INVITE messages and 
should use a new From tag.

Corrected Text
--------------
message 1
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
   Supported: timer
   Session-Expires: 50
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp
   Content-Length: 142

message 4
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
   Supported: timer
   Session-Expires: 3600
   Min-SE: 3600
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=2568701785
   Call-ID: a84b4c76e66710
   CSeq: 314160 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp
   Content-Length: 142

message 10
   INVITE sips:bob@biloxi.example.com SIP/2.0
   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
   Supported: timer
   Session-Expires: 4000
   Min-SE: 4000
   Max-Forwards: 70
   To: Bob <sips:bob@biloxi.example.com>
   From: Alice <sips:alice@atlanta.example.com>;tag=5647301796
   Call-ID: a84b4c76e66710
   CSeq: 314161 INVITE
   Contact: <sips:alice@pc33.atlanta.example.com>
   Content-Type: application/sdp

Notes
-----
-----Original Message-----
From: Paul Kyzivat (pkyzivat) 
Sent: Monday, February 09, 2009 10:09 PM
To: Muthu ArulMozhi Perumal (mperumal)
Cc: Radha Krishna Saragadam (rsaragad); Jonathan Rosenberg (jdrosen); Ram Mohan R (rmohanr)
Subject: Re: UAS behavior after sending 422 for initial INVITE

yes, I think so.

	Paul

Muthu ArulMozhi Perumal (mperumal) wrote:
> In section 13 (Example Call Flow) of RFC 4028 the From tag never changes
> between the initial INVITE message and the subsequent INVITE messages
> sent after receiving a 422:
> 
> message 1
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>    Call-ID: a84b4c76e66710
> 
> message 4
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>    Call-ID: a84b4c76e66710
> 
> message 10
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>    Call-ID: a84b4c76e66710
> 
> Is this a bug in the RFC?
> 
> thanks,
> Muthu
> 
> |-----Original Message-----
> |From: Paul Kyzivat (pkyzivat)
> |Sent: Wednesday, February 04, 2009 12:36 AM
> |To: Radha Krishna Saragadam (rsaragad)
> |Cc: Jonathan Rosenberg (jdrosen); Muthu ArulMozhi Perumal (mperumal);
> Ram Mohan R (rmohanr)
> |Subject: Re: UAS behavior after sending 422 for initial INVITE
> |
> |Radha,
> |
> |It is not a reinvite, because a dialog was never established - the
> first
> |call failed.
> |
> |So you are starting a new invite. You can use the same callid, but
> |should use a new from-tag.
> |
> |	Thanks,
> |	Paul
> |
> |Radha Krishna Saragadam (rsaragad) wrote:
> |> Hi Paul
> |>
> |> 	My question is for initial INVITE. For initial INVITE if UA
> |> receives 422 and UA want to retry INVITE with new value increased
> value
> |> then what should be the To(with tag), From(with tag) and CallID
> values?
> |> Is it a Re-INVITE or new a Dialog? Section 7.3 says same value for
> |> To,From and CallID
> |>
> |> Regards
> |> S.Radha krishna

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
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can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC4028 (draft-ietf-sip-session-timer-15)
--------------------------------------
Title               : Session Timers in the Session Initiation Protocol (SIP)
Publication Date    : April 2005
Author(s)           : S. Donovan, J. Rosenberg
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From brett@broadsoft.com  Mon Feb  9 09:59:49 2009
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From: Brett Tate <brett@broadsoft.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Mon, 9 Feb 2009 09:59:46 -0800
Thread-Topic: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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The reported error does not sound correct.  If RFC 4028 is incorrect concer=
ning From tag, so is RFC 3261 (section 8.1 and likely other sections) and v=
arious other RFCs.



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of RFC
> Errata System
> Sent: Monday, February 09, 2009 12:37 PM
> To: srd@cisco.com; jdrosen@cisco.com; jon.peterson@neustar.biz;
> fluffy@cisco.com; dean.willis@softarmor.com; drage@alcatel-lucent.com
> Cc: sip@ietf.org; mperumal@cisco.com; rfc-editor@rfc-editor.org
> Subject: [Sip] [Technical Errata Reported] RFC4028 (1681)
>=20
>=20
> The following errata report has been submitted for RFC4028,
> "Session Timers in the Session Initiation Protocol (SIP)".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D4028&eid=3D1681
>=20
> --------------------------------------
> Type: Technical
> Reported by: Muthu Arul Mozhi <mperumal@cisco.com>
>=20
> Section: 13
>=20
> Original Text
> -------------
> In section 13 (Example Call Flow) the From tag never changes
> between the initial INVITE message and the subsequent INVITE
> messages sent after receiving a 422:
>=20
> message 1
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds8
>    Supported: timer
>    Session-Expires: 50
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
>    Call-ID: a84b4c76e66710
>    CSeq: 314159 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>    Content-Length: 142
>=20
> message 4
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds9
>    Supported: timer
>    Session-Expires: 3600
>    Min-SE: 3600
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
>    Call-ID: a84b4c76e66710
>    CSeq: 314160 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>    Content-Length: 142
>=20
> message 10
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds10
>    Supported: timer
>    Session-Expires: 4000
>    Min-SE: 4000
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
>    Call-ID: a84b4c76e66710
>    CSeq: 314161 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>=20
> However, as per RFC 3261, if an initial INVITE generates a non-2xx final
> response, that terminates all sessions and all dialogs that were created.
>=20
> Hence, these are not re-INVITE messages, rather new INVITE messages and
> should use a new From tag.
>=20
> Corrected Text
> --------------
> message 1
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds8
>    Supported: timer
>    Session-Expires: 50
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
>    Call-ID: a84b4c76e66710
>    CSeq: 314159 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>    Content-Length: 142
>=20
> message 4
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds9
>    Supported: timer
>    Session-Expires: 3600
>    Min-SE: 3600
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D2568701785
>    Call-ID: a84b4c76e66710
>    CSeq: 314160 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>    Content-Length: 142
>=20
> message 10
>    INVITE sips:bob@biloxi.example.com SIP/2.0
>    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds10
>    Supported: timer
>    Session-Expires: 4000
>    Min-SE: 4000
>    Max-Forwards: 70
>    To: Bob <sips:bob@biloxi.example.com>
>    From: Alice <sips:alice@atlanta.example.com>;tag=3D5647301796
>    Call-ID: a84b4c76e66710
>    CSeq: 314161 INVITE
>    Contact: <sips:alice@pc33.atlanta.example.com>
>    Content-Type: application/sdp
>=20
> Notes
> -----
> -----Original Message-----
> From: Paul Kyzivat (pkyzivat)
> Sent: Monday, February 09, 2009 10:09 PM
> To: Muthu ArulMozhi Perumal (mperumal)
> Cc: Radha Krishna Saragadam (rsaragad); Jonathan Rosenberg (jdrosen); Ram
> Mohan R (rmohanr)
> Subject: Re: UAS behavior after sending 422 for initial INVITE
>=20
> yes, I think so.
>=20
> 	Paul
>=20
> Muthu ArulMozhi Perumal (mperumal) wrote:
> > In section 13 (Example Call Flow) of RFC 4028 the From tag never change=
s
> > between the initial INVITE message and the subsequent INVITE messages
> > sent after receiving a 422:
> >
> > message 1
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >
> > message 4
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >
> > message 10
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >
> > Is this a bug in the RFC?
> >
> > thanks,
> > Muthu
> >
> > |-----Original Message-----
> > |From: Paul Kyzivat (pkyzivat)
> > |Sent: Wednesday, February 04, 2009 12:36 AM
> > |To: Radha Krishna Saragadam (rsaragad)
> > |Cc: Jonathan Rosenberg (jdrosen); Muthu ArulMozhi Perumal (mperumal);
> > Ram Mohan R (rmohanr)
> > |Subject: Re: UAS behavior after sending 422 for initial INVITE
> > |
> > |Radha,
> > |
> > |It is not a reinvite, because a dialog was never established - the
> > first
> > |call failed.
> > |
> > |So you are starting a new invite. You can use the same callid, but
> > |should use a new from-tag.
> > |
> > |	Thanks,
> > |	Paul
> > |
> > |Radha Krishna Saragadam (rsaragad) wrote:
> > |> Hi Paul
> > |>
> > |> 	My question is for initial INVITE. For initial INVITE if UA
> > |> receives 422 and UA want to retry INVITE with new value increased
> > value
> > |> then what should be the To(with tag), From(with tag) and CallID
> > values?
> > |> Is it a Re-INVITE or new a Dialog? Section 7.3 says same value for
> > |> To,From and CallID
> > |>
> > |> Regards
> > |> S.Radha krishna
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>=20
> --------------------------------------
> RFC4028 (draft-ietf-sip-session-timer-15)
> --------------------------------------
> Title               : Session Timers in the Session Initiation Protocol
> (SIP)
> Publication Date    : April 2005
> Author(s)           : S. Donovan, J. Rosenberg
> Category            : PROPOSED STANDARD
> Source              : Session Initiation Protocol
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> _______________________________________________
> Sip mailing list  https://www.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 brett@broadsoft.com  Mon Feb  9 10:03:18 2009
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From: Brett Tate <brett@broadsoft.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Mon, 9 Feb 2009 10:03:21 -0800
Thread-Topic: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Sorry about the repost, I intended to indicate section RFC 3261 section 8.1=
.3.5.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Brett Tate
> Sent: Monday, February 09, 2009 1:00 PM
> To: sip@ietf.org
> Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>=20
> The reported error does not sound correct.  If RFC 4028 is incorrect
> concerning From tag, so is RFC 3261 (section 8.1 and likely other
> sections) and various other RFCs.
>=20
>=20
>=20
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> RFC
> > Errata System
> > Sent: Monday, February 09, 2009 12:37 PM
> > To: srd@cisco.com; jdrosen@cisco.com; jon.peterson@neustar.biz;
> > fluffy@cisco.com; dean.willis@softarmor.com; drage@alcatel-lucent.com
> > Cc: sip@ietf.org; mperumal@cisco.com; rfc-editor@rfc-editor.org
> > Subject: [Sip] [Technical Errata Reported] RFC4028 (1681)
> >
> >
> > The following errata report has been submitted for RFC4028,
> > "Session Timers in the Session Initiation Protocol (SIP)".
> >
> > --------------------------------------
> > You may review the report below and at:
> > http://www.rfc-editor.org/errata_search.php?rfc=3D4028&eid=3D1681
> >
> > --------------------------------------
> > Type: Technical
> > Reported by: Muthu Arul Mozhi <mperumal@cisco.com>
> >
> > Section: 13
> >
> > Original Text
> > -------------
> > In section 13 (Example Call Flow) the From tag never changes
> > between the initial INVITE message and the subsequent INVITE
> > messages sent after receiving a 422:
> >
> > message 1
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds8
> >    Supported: timer
> >    Session-Expires: 50
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314159 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >    Content-Length: 142
> >
> > message 4
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds9
> >    Supported: timer
> >    Session-Expires: 3600
> >    Min-SE: 3600
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314160 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >    Content-Length: 142
> >
> > message 10
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds10
> >    Supported: timer
> >    Session-Expires: 4000
> >    Min-SE: 4000
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314161 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >
> > However, as per RFC 3261, if an initial INVITE generates a non-2xx fina=
l
> > response, that terminates all sessions and all dialogs that were
> created.
> >
> > Hence, these are not re-INVITE messages, rather new INVITE messages and
> > should use a new From tag.
> >
> > Corrected Text
> > --------------
> > message 1
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds8
> >    Supported: timer
> >    Session-Expires: 50
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314159 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >    Content-Length: 142
> >
> > message 4
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds9
> >    Supported: timer
> >    Session-Expires: 3600
> >    Min-SE: 3600
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D2568701785
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314160 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >    Content-Length: 142
> >
> > message 10
> >    INVITE sips:bob@biloxi.example.com SIP/2.0
> >    Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=3Dz9hG4bKnashds10
> >    Supported: timer
> >    Session-Expires: 4000
> >    Min-SE: 4000
> >    Max-Forwards: 70
> >    To: Bob <sips:bob@biloxi.example.com>
> >    From: Alice <sips:alice@atlanta.example.com>;tag=3D5647301796
> >    Call-ID: a84b4c76e66710
> >    CSeq: 314161 INVITE
> >    Contact: <sips:alice@pc33.atlanta.example.com>
> >    Content-Type: application/sdp
> >
> > Notes
> > -----
> > -----Original Message-----
> > From: Paul Kyzivat (pkyzivat)
> > Sent: Monday, February 09, 2009 10:09 PM
> > To: Muthu ArulMozhi Perumal (mperumal)
> > Cc: Radha Krishna Saragadam (rsaragad); Jonathan Rosenberg (jdrosen);
> Ram
> > Mohan R (rmohanr)
> > Subject: Re: UAS behavior after sending 422 for initial INVITE
> >
> > yes, I think so.
> >
> > 	Paul
> >
> > Muthu ArulMozhi Perumal (mperumal) wrote:
> > > In section 13 (Example Call Flow) of RFC 4028 the From tag never
> changes
> > > between the initial INVITE message and the subsequent INVITE messages
> > > sent after receiving a 422:
> > >
> > > message 1
> > >    INVITE sips:bob@biloxi.example.com SIP/2.0
> > >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> > >    Call-ID: a84b4c76e66710
> > >
> > > message 4
> > >    INVITE sips:bob@biloxi.example.com SIP/2.0
> > >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> > >    Call-ID: a84b4c76e66710
> > >
> > > message 10
> > >    INVITE sips:bob@biloxi.example.com SIP/2.0
> > >    From: Alice <sips:alice@atlanta.example.com>;tag=3D1928301774
> > >    Call-ID: a84b4c76e66710
> > >
> > > Is this a bug in the RFC?
> > >
> > > thanks,
> > > Muthu
> > >
> > > |-----Original Message-----
> > > |From: Paul Kyzivat (pkyzivat)
> > > |Sent: Wednesday, February 04, 2009 12:36 AM
> > > |To: Radha Krishna Saragadam (rsaragad)
> > > |Cc: Jonathan Rosenberg (jdrosen); Muthu ArulMozhi Perumal (mperumal)=
;
> > > Ram Mohan R (rmohanr)
> > > |Subject: Re: UAS behavior after sending 422 for initial INVITE
> > > |
> > > |Radha,
> > > |
> > > |It is not a reinvite, because a dialog was never established - the
> > > first
> > > |call failed.
> > > |
> > > |So you are starting a new invite. You can use the same callid, but
> > > |should use a new from-tag.
> > > |
> > > |	Thanks,
> > > |	Paul
> > > |
> > > |Radha Krishna Saragadam (rsaragad) wrote:
> > > |> Hi Paul
> > > |>
> > > |> 	My question is for initial INVITE. For initial INVITE if UA
> > > |> receives 422 and UA want to retry INVITE with new value increased
> > > value
> > > |> then what should be the To(with tag), From(with tag) and CallID
> > > values?
> > > |> Is it a Re-INVITE or new a Dialog? Section 7.3 says same value for
> > > |> To,From and CallID
> > > |>
> > > |> Regards
> > > |> S.Radha krishna
> >
> > Instructions:
> > -------------
> > This errata is currently posted as "Reported". If necessary, please
> > use "Reply All" to discuss whether it should be verified or
> > rejected. When a decision is reached, the verifying party (IESG)
> > can log in to change the status and edit the report, if necessary.
> >
> > --------------------------------------
> > RFC4028 (draft-ietf-sip-session-timer-15)
> > --------------------------------------
> > Title               : Session Timers in the Session Initiation Protocol
> > (SIP)
> > Publication Date    : April 2005
> > Author(s)           : S. Donovan, J. Rosenberg
> > Category            : PROPOSED STANDARD
> > Source              : Session Initiation Protocol
> > Area                : Real-time Applications and Infrastructure
> > Stream              : IETF
> > Verifying Party     : IESG
> > _______________________________________________
> > Sip mailing list  https://www.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://www.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 rjsparks@nostrum.com  Mon Feb  9 11:03:52 2009
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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I agree - the example is not flawed (at least in the way the errata  
reports).

Muthu seems to troubled by the reuse of the call-id and from tag when  
the initial transaction didn't create a dialog.
While doing so is not required by the specification, nothing makes it  
illegal either.

RjS

On Feb 9, 2009, at 11:59 AM, Brett Tate wrote:

> The reported error does not sound correct.  If RFC 4028 is incorrect  
> concerning From tag, so is RFC 3261 (section 8.1 and likely other  
> sections) and various other RFCs.
>
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf  
>> Of RFC
>> Errata System
>> Sent: Monday, February 09, 2009 12:37 PM
>> To: srd@cisco.com; jdrosen@cisco.com; jon.peterson@neustar.biz;
>> fluffy@cisco.com; dean.willis@softarmor.com; drage@alcatel-lucent.com
>> Cc: sip@ietf.org; mperumal@cisco.com; rfc-editor@rfc-editor.org
>> Subject: [Sip] [Technical Errata Reported] RFC4028 (1681)
>>
>>
>> The following errata report has been submitted for RFC4028,
>> "Session Timers in the Session Initiation Protocol (SIP)".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=4028&eid=1681
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Muthu Arul Mozhi <mperumal@cisco.com>
>>
>> Section: 13
>>
>> Original Text
>> -------------
>> In section 13 (Example Call Flow) the From tag never changes
>> between the initial INVITE message and the subsequent INVITE
>> messages sent after receiving a 422:
>>
>> message 1
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
>>   Supported: timer
>>   Session-Expires: 50
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314159 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>   Content-Length: 142
>>
>> message 4
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
>>   Supported: timer
>>   Session-Expires: 3600
>>   Min-SE: 3600
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314160 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>   Content-Length: 142
>>
>> message 10
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
>>   Supported: timer
>>   Session-Expires: 4000
>>   Min-SE: 4000
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314161 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>
>> However, as per RFC 3261, if an initial INVITE generates a non-2xx  
>> final
>> response, that terminates all sessions and all dialogs that were  
>> created.
>>
>> Hence, these are not re-INVITE messages, rather new INVITE messages  
>> and
>> should use a new From tag.
>>
>> Corrected Text
>> --------------
>> message 1
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds8
>>   Supported: timer
>>   Session-Expires: 50
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314159 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>   Content-Length: 142
>>
>> message 4
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds9
>>   Supported: timer
>>   Session-Expires: 3600
>>   Min-SE: 3600
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=2568701785
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314160 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>   Content-Length: 142
>>
>> message 10
>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>   Via: SIP/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bKnashds10
>>   Supported: timer
>>   Session-Expires: 4000
>>   Min-SE: 4000
>>   Max-Forwards: 70
>>   To: Bob <sips:bob@biloxi.example.com>
>>   From: Alice <sips:alice@atlanta.example.com>;tag=5647301796
>>   Call-ID: a84b4c76e66710
>>   CSeq: 314161 INVITE
>>   Contact: <sips:alice@pc33.atlanta.example.com>
>>   Content-Type: application/sdp
>>
>> Notes
>> -----
>> -----Original Message-----
>> From: Paul Kyzivat (pkyzivat)
>> Sent: Monday, February 09, 2009 10:09 PM
>> To: Muthu ArulMozhi Perumal (mperumal)
>> Cc: Radha Krishna Saragadam (rsaragad); Jonathan Rosenberg  
>> (jdrosen); Ram
>> Mohan R (rmohanr)
>> Subject: Re: UAS behavior after sending 422 for initial INVITE
>>
>> yes, I think so.
>>
>> 	Paul
>>
>> Muthu ArulMozhi Perumal (mperumal) wrote:
>>> In section 13 (Example Call Flow) of RFC 4028 the From tag never  
>>> changes
>>> between the initial INVITE message and the subsequent INVITE  
>>> messages
>>> sent after receiving a 422:
>>>
>>> message 1
>>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>>   Call-ID: a84b4c76e66710
>>>
>>> message 4
>>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>>   Call-ID: a84b4c76e66710
>>>
>>> message 10
>>>   INVITE sips:bob@biloxi.example.com SIP/2.0
>>>   From: Alice <sips:alice@atlanta.example.com>;tag=1928301774
>>>   Call-ID: a84b4c76e66710
>>>
>>> Is this a bug in the RFC?
>>>
>>> thanks,
>>> Muthu
>>>
>>> |-----Original Message-----
>>> |From: Paul Kyzivat (pkyzivat)
>>> |Sent: Wednesday, February 04, 2009 12:36 AM
>>> |To: Radha Krishna Saragadam (rsaragad)
>>> |Cc: Jonathan Rosenberg (jdrosen); Muthu ArulMozhi Perumal  
>>> (mperumal);
>>> Ram Mohan R (rmohanr)
>>> |Subject: Re: UAS behavior after sending 422 for initial INVITE
>>> |
>>> |Radha,
>>> |
>>> |It is not a reinvite, because a dialog was never established - the
>>> first
>>> |call failed.
>>> |
>>> |So you are starting a new invite. You can use the same callid, but
>>> |should use a new from-tag.
>>> |
>>> |	Thanks,
>>> |	Paul
>>> |
>>> |Radha Krishna Saragadam (rsaragad) wrote:
>>> |> Hi Paul
>>> |>
>>> |> 	My question is for initial INVITE. For initial INVITE if UA
>>> |> receives 422 and UA want to retry INVITE with new value increased
>>> value
>>> |> then what should be the To(with tag), From(with tag) and CallID
>>> values?
>>> |> Is it a Re-INVITE or new a Dialog? Section 7.3 says same value  
>>> for
>>> |> To,From and CallID
>>> |>
>>> |> Regards
>>> |> S.Radha krishna
>>
>> Instructions:
>> -------------
>> This errata is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC4028 (draft-ietf-sip-session-timer-15)
>> --------------------------------------
>> Title               : Session Timers in the Session Initiation  
>> Protocol
>> (SIP)
>> Publication Date    : April 2005
>> Author(s)           : S. Donovan, J. Rosenberg
>> Category            : PROPOSED STANDARD
>> Source              : Session Initiation Protocol
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>> _______________________________________________
>> Sip mailing list  https://www.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://www.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 dworley@nortel.com  Mon Feb  9 13:14:40 2009
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:
> I agree - the example is not flawed (at least in the way the errata  
> reports).
> 
> Muthu seems to troubled by the reuse of the call-id and from tag when  
> the initial transaction didn't create a dialog.
> While doing so is not required by the specification, nothing makes it  
> illegal either.

Are we agreed that "doing so is not required by the specification"?  In
RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and
420 responses are required to have the same from-tag as the original
request.  Is 422 different?

Dale



From brett@broadsoft.com  Mon Feb  9 13:25:06 2009
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To: Dale Worley <dworley@nortel.com>, SIP <sip@ietf.org>
Date: Mon, 9 Feb 2009 13:25:06 -0800
Thread-Topic: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Based upon the following snippet, my understanding is that RFC 4028 is corr=
ectly following the SHOULD.

RFC 3261 section 8.1.3.5 snippet:

"In all of the above cases, the request is retried by creating a new reques=
t with the appropriate modifications.  This new request constitutes a new t=
ransaction and SHOULD have the same value of the Call-ID, To, and From of t=
he previous request, but the CSeq should contain a new sequence number that=
 is one higher than the previous. With other 4xx responses, including those=
 yet to be defined, a retry may or may not be possible depending on the met=
hod and the use case."

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e
> Worley
> Sent: Monday, February 09, 2009 4:14 PM
> To: SIP
> Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>=20
> On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:
> > I agree - the example is not flawed (at least in the way the errata
> > reports).
> >
> > Muthu seems to troubled by the reuse of the call-id and from tag when
> > the initial transaction didn't create a dialog.
> > While doing so is not required by the specification, nothing makes it
> > illegal either.
>=20
> Are we agreed that "doing so is not required by the specification"?  In
> RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and
> 420 responses are required to have the same from-tag as the original
> request.  Is 422 different?
>=20
> Dale
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www.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 dworley@nortel.com  Mon Feb  9 15:04:13 2009
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From: "Dale Worley" <dworley@nortel.com>
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Subject: [Sip] Retargeting difficulties
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It is my contention that the "retargeting problem" is more complex than
people have been allowing for.  In examples that have been presented,
there have usually been at most two relevant retarget/redirect
operations, and all of them have been associated with one single AOR.

For example, in section 2.1 of
draft-rosenberg-sip-target-uri-delivery-00, "Unknown Aliases", the
situation is:

sip:worley-alias@serviceprovider ->  (alias address)
     sip:worley@serviceprovider ->   (AOR)
          sip:10.1.1.123             (registered contact)

Both of these mappings are tightly coupled with AOR
sip:worley@serviceprovider.

But let us consider a more complicated example:

I am included as an element of a "hunt group" for some work-related
purpose:  sip:hunt@group -> sip:worley@serviceprovider

I have a UA that registers:  sip:worley@serviceprovider -> sip:10.1.1.1

However, I am vacationing at a friend's Roman villa, and since I haven't
paid serviceprovider for call forwarding, I direct my home UA to forward
my calls:  sip:10.1.1.1 -> sip:amico@fornitorediservizio

I take another UA to the Roman villa, where it registers:
sip:amico@fornitorediservizio -> sip:10.2.2.2

sip:hunt@group ->
     sip:worley@serviceprovider ->
          sip:10.1.1.1 ->
               sip:amico@fornitorediservizio ->
                    sip:10.2.2.2

Each of these mappings is associated with a different
business/social/technological activity.

Now let us assume that a call initially directed to sip:hunt@group
arrives as sip:10.2.2.2.  What should my UA do with it?  Or rather,
which request-URI should it consider to be the "real" target of the
call?

Once you look at it, the answer clearly depends on which of these URIs
the UA has knowledge of.  If it is stupid, all it can understand is that
there is an incoming call, and alert me.  If it has some knowledge about
the AOR sip:amico@fornitorediservizio, it can base its action on the
fact that the call came to that AOR, or in variants of this scenario,
that the call came via a request-URI that was related to this AOR.

If the phone understands that sip:worley@serviceprovider has been
forwarded to sip:amico@fornitorediservizio, it can alert in a different
manner than it would for calls that are "just" to
sip:amico@fornitorediservizio (*).  And if the phone knows about
sip:hunt@group, it could provide yet another alternative action.

So I think that any solution to the "retargeting problem" that is going
to be practical in the long run -- that is, will be adaptable to future
networks where multiple "features" are ganged together -- because it is
the only solution that has "recursiveness", "If you can do it
[retargeting] once, you can do it many times."  Recursiveness requires
that the entire request-URI history of the call be presented to the UAS,
and that the UAS will inspect that history for the earliest request-URI
that it has knowledge of to determine the "context" that it should apply
to this request.

Fortunately, History-Info is (or is close to being) the needed solution.

Conversely, all other proposed solutions fail to provide enough
information:  They either provide the *next-to-last* request-URI
(P-Called-Party-Id) or the *first* request-URI (From or Target header).
This data is insufficient, as can be seen from the carefully constructed
example (*) above -- where the UAS does not understand the first
request-URI, and the last and next-to-last request-URIs do not contain
the information the UAS should act on -- only the 2nd of the 5
request-URIs contains the needed information.

Dale



From muthu.arul@gmail.com  Mon Feb  9 20:57:20 2009
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Date: Tue, 10 Feb 2009 10:27:22 +0530
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From: Muthu Arul <muthu.arul@gmail.com>
To: Brett Tate <brett@broadsoft.com>
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Cc: SIP <sip@ietf.org>
Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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If the From tag doesn't change, can the UAS retrain the same To tag as well?
If the new INVITE reaches the UAS before the ACK for the previous INVITE,
what should the UAS do? Should it reject the new INVITE and expect the UAC
to retry again?

I thought it would have been much simpler had the From/To tag changed b/w
the INVITE messages..

thanks,
Muthu

On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <brett@broadsoft.com> wrote:

> Based upon the following snippet, my understanding is that RFC 4028 is
> correctly following the SHOULD.
>
> RFC 3261 section 8.1.3.5 snippet:
>
> "In all of the above cases, the request is retried by creating a new
> request with the appropriate modifications.  This new request constitutes a
> new transaction and SHOULD have the same value of the Call-ID, To, and From
> of the previous request, but the CSeq should contain a new sequence number
> that is one higher than the previous. With other 4xx responses, including
> those yet to be defined, a retry may or may not be possible depending on the
> method and the use case."
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Dale
> > Worley
> > Sent: Monday, February 09, 2009 4:14 PM
> > To: SIP
> > Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
> >
> > On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:
> > > I agree - the example is not flawed (at least in the way the errata
> > > reports).
> > >
> > > Muthu seems to troubled by the reuse of the call-id and from tag when
> > > the initial transaction didn't create a dialog.
> > > While doing so is not required by the specification, nothing makes it
> > > illegal either.
> >
> > Are we agreed that "doing so is not required by the specification"?  In
> > RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and
> > 420 responses are required to have the same from-tag as the original
> > request.  Is 422 different?
> >
> > Dale
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www.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://www.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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If the From tag doesn&#39;t change, can the UAS retrain the same To tag as =
well? If the new INVITE reaches the UAS before the ACK for the previous INV=
ITE, what should the UAS do? Should it reject the new INVITE and expect the=
 UAC to retry again?<br>
<br>I thought it would have been much simpler had the From/To tag changed b=
/w the INVITE messages..<br><br>thanks,<br>Muthu<br><br><div class=3D"gmail=
_quote">On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <span dir=3D"ltr">&lt;<=
a href=3D"mailto:brett@broadsoft.com">brett@broadsoft.com</a>&gt;</span> wr=
ote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Based upon the fo=
llowing snippet, my understanding is that RFC 4028 is correctly following t=
he SHOULD.<br>

<br>
RFC 3261 section 8.1.3.5 snippet:<br>
<br>
&quot;In all of the above cases, the request is retried by creating a new r=
equest with the appropriate modifications. &nbsp;This new request constitut=
es a new transaction and SHOULD have the same value of the Call-ID, To, and=
 From of the previous request, but the CSeq should contain a new sequence n=
umber that is one higher than the previous. With other 4xx responses, inclu=
ding those yet to be defined, a retry may or may not be possible depending =
on the method and the use case.&quot;<br>

<div><div></div><div class=3D"Wj3C7c"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>=
 [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] =
On Behalf Of Dale<br>
&gt; Worley<br>
&gt; Sent: Monday, February 09, 2009 4:14 PM<br>
&gt; To: SIP<br>
&gt; Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)<br>
&gt;<br>
&gt; On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:<br>
&gt; &gt; I agree - the example is not flawed (at least in the way the erra=
ta<br>
&gt; &gt; reports).<br>
&gt; &gt;<br>
&gt; &gt; Muthu seems to troubled by the reuse of the call-id and from tag =
when<br>
&gt; &gt; the initial transaction didn&#39;t create a dialog.<br>
&gt; &gt; While doing so is not required by the specification, nothing make=
s it<br>
&gt; &gt; illegal either.<br>
&gt;<br>
&gt; Are we agreed that &quot;doing so is not required by the specification=
&quot;? &nbsp;In<br>
&gt; RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and=
<br>
&gt; 420 responses are required to have the same from-tag as the original<b=
r>
&gt; request. &nbsp;Is 422 different?<br>
&gt;<br>
&gt; Dale<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinf=
o/sip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implemento=
rs@cs.columbia.edu</a> for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new d=
evelopments on the application of sip<br>
_______________________________________________<br>
Sip mailing list &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
This list is for NEW development of the core SIP Protocol<br>
Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs=
.columbia.edu</a> for questions on current sip<br>
Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develo=
pments on the application of sip<br>
</div></div></blockquote></div><br>

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Subject: [Sip] I-D Action:draft-ietf-sip-ua-privacy-05.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : UA-Driven Privacy Mechanism for SIP
	Author(s)       : M. Munakata, et al.
	Filename        : draft-ietf-sip-ua-privacy-05.txt
	Pages           : 11
	Date            : 2009-02-09

This document defines a guideline for a user agent to generate an
anonymous SIP message by utilizing mechanisms such as GRUU (Globally
Routable User Agent URIs) and TURN (Traversal Using Relays around
NAT) without the need for a privacy service defined in RFC 3323.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-05.txt

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From munakata.mayumi@lab.ntt.co.jp  Mon Feb  9 21:20:01 2009
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From: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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Subject: [Sip] Update of draft-ietf-sip-ua-privacy
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All;

I have updated the ua-privacy draft.
http://www.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-05.txt

After submission of -04 version, John pointed that
some nits comments were not taken into account.
So I submitted -05 version to incorporate them.

Changes between -03 and -05 are as follows:

Krishna and John gave us valuable comments in
mail list and privately (Thank you to both of you!)
and I incorporated the comments as follows;

1) In Section 4.1,
mentions that third party GRUU server is out of scope.

2) In Section 5.1.2,
mentions that third party anonymization service is out of scope.

3) In Section 5.1.1,
deleted the use of TURN mechanism for the Contact header.

4) In Section 5.1.1,
changed the text in order not to anonymize the Contact header
in REGISTER requests and 3xx responses.

5) Many editorial changes that John proposed.

I also changed the expression of "best current practice"
to "guideline" because people agreed that this document
should be Informational not BCP.

The editors believe that there is no remaining issues.

Thanks,
Mayumi

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References: <200902091736.n19HaYee027174@boreas.isi.edu><9B2A061A1137254BBE4F4B2CD843646A0FD3B1DE5F@mbx02.citservers.local><A560AE7E-8E26-4D6D-B487-B98A3EDAB964@nostrum.com><1234214049.4033.62.camel@victoria-pingtel-com.us.nortel.com><9B2A061A1137254BBE4F4B2CD843646A0FD3B1DF1E@mbx02.citservers.local> <8c400f2e0902092057y5b98ce09nc69e5748d6032faa@mail.gmail.com>
From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Muthu Arul" <muthu.arul@gmail.com>, "Brett Tate" <brett@broadsoft.com>
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Pl. see inline ...


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Muthu Arul
	Sent: Tuesday, February 10, 2009 10:27 AM
	To: Brett Tate
	Cc: SIP
	Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
=09
=09
	If the From tag doesn't change, can the UAS retrain the same To
tag as well?=20
	[Sanjay Sinha (sanjsinh)]  When can UAS retain to-tag, when it
is rejecting *resubmitted* request?
	=20
	 If the new INVITE reaches the UAS before the ACK for the
previous INVITE, what should the UAS do?=20
	[Sanjay Sinha (sanjsinh)] Since the resubmitted request will not
have a to-tag, even if Call-Id and from-tag is same, it wont match any
ongoing transaction and so should be treated as a new request.=20
	=20
	Should it reject the new INVITE and expect the UAC to retry
again?
=09
	I thought it would have been much simpler had the From/To tag
changed b/w the INVITE messages..
	[Sanjay Sinha (sanjsinh)]  It would have been better, but it is
not illegal for Call-id + from-tag to not change. Besides, it is an
example and should be treated as such..=20
=09
	thanks,
	Muthu
=09
=09
	On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate
<brett@broadsoft.com> wrote:
=09

		Based upon the following snippet, my understanding is
that RFC 4028 is correctly following the SHOULD.
	=09
		RFC 3261 section 8.1.3.5 snippet:
	=09
		"In all of the above cases, the request is retried by
creating a new request with the appropriate modifications.  This new
request constitutes a new transaction and SHOULD have the same value of
the Call-ID, To, and From of the previous request, but the CSeq should
contain a new sequence number that is one higher than the previous. With
other 4xx responses, including those yet to be defined, a retry may or
may not be possible depending on the method and the use case."
	=09

		> -----Original Message-----
		> From: sip-bounces@ietf.org
[mailto:sip-bounces@ietf.org] On Behalf Of Dale
		> Worley
		> Sent: Monday, February 09, 2009 4:14 PM
		> To: SIP
		> Subject: Re: [Sip] [Technical Errata Reported] RFC4028
(1681)
		>
		> On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks
wrote:
		> > I agree - the example is not flawed (at least in the
way the errata
		> > reports).
		> >
		> > Muthu seems to troubled by the reuse of the call-id
and from tag when
		> > the initial transaction didn't create a dialog.
		> > While doing so is not required by the specification,
nothing makes it
		> > illegal either.
		>
		> Are we agreed that "doing so is not required by the
specification"?  In
		> RFC 3261, requests that are retried after 401, 407,
413, 415, 416, and
		> 420 responses are required to have the same from-tag
as the original
		> request.  Is 422 different?
		>
		> Dale
		>
		>
		> _______________________________________________
		> Sip mailing list
https://www.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://www.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
	=09



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charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D420460508-10022009></SPAN><FONT=20
face=3DArial><FONT color=3D#0000ff><FONT size=3D2>P<SPAN =
class=3D420460508-10022009>l.=20
see inline ...</SPAN></FONT></FONT></FONT><BR></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Muthu =
Arul<BR><B>Sent:</B>=20
  Tuesday, February 10, 2009 10:27 AM<BR><B>To:</B> Brett =
Tate<BR><B>Cc:</B>=20
  SIP<BR><B>Subject:</B> Re: [Sip] [Technical Errata Reported] RFC4028=20
  (1681)<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV>If the From tag doesn't change, can the UAS retrain the same To =
tag as=20
  well? <BR><SPAN class=3D420460508-10022009><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>[Sanjay Sinha (sanjsinh)]&nbsp;&nbsp;When&nbsp;can UAS retain =
to-tag,=20
  when it is rejecting *resubmitted* request?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D420460508-10022009></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D420460508-10022009></SPAN><SPAN=20
  class=3D420460508-10022009>&nbsp;</SPAN>If the new INVITE reaches the =
UAS before=20
  the ACK for the previous INVITE, what should the UAS do? <BR><SPAN=20
  class=3D420460508-10022009><FONT face=3DArial color=3D#0000ff =
size=3D2>[Sanjay Sinha=20
  (sanjsinh)]&nbsp;Since the resubmitted request will not have a to-tag, =

  even&nbsp;if Call-Id and from-tag is same,&nbsp;it wont match any =
ongoing=20
  transaction and so should be treated as a new=20
  request.&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=3D420460508-10022009><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV>Should it reject the new INVITE and expect the UAC to retry=20
  again?<BR><BR>I thought it would have been much simpler had the =
From/To tag=20
  changed b/w the INVITE messages..<BR><SPAN =
class=3D420460508-10022009><FONT=20
  face=3DArial color=3D#0000ff size=3D2>[Sanjay Sinha (sanjsinh)]&nbsp; =
It&nbsp;would=20
  have&nbsp;been better, but it is not illegal for Call-id + from-tag to =

  not&nbsp;change. Besides, it is an example and should be treated as=20
  such..&nbsp;</FONT></SPAN><BR><BR>thanks,<BR>Muthu<BR><BR></DIV>
  <DIV class=3Dgmail_quote>On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate =
<SPAN=20
  dir=3Dltr>&lt;<A=20
  href=3D"mailto:brett@broadsoft.com">brett@broadsoft.com</A>&gt;</SPAN> =

wrote:<BR>
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: =
rgb(204,204,204) 1px solid">Based=20
    upon the following snippet, my understanding is that RFC 4028 is =
correctly=20
    following the SHOULD.<BR><BR>RFC 3261 section 8.1.3.5 =
snippet:<BR><BR>"In=20
    all of the above cases, the request is retried by creating a new =
request=20
    with the appropriate modifications. &nbsp;This new request =
constitutes a new=20
    transaction and SHOULD have the same value of the Call-ID, To, and =
>From of=20
    the previous request, but the CSeq should contain a new sequence =
number that=20
    is one higher than the previous. With other 4xx responses, including =
those=20
    yet to be defined, a retry may or may not be possible depending on =
the=20
    method and the use case."<BR>
    <DIV>
    <DIV></DIV>
    <DIV class=3DWj3C7c><BR>&gt; -----Original Message-----<BR>&gt; =
From: <A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A> =
[mailto:<A=20
    href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>] On =
Behalf Of=20
    Dale<BR>&gt; Worley<BR>&gt; Sent: Monday, February 09, 2009 4:14 =
PM<BR>&gt;=20
    To: SIP<BR>&gt; Subject: Re: [Sip] [Technical Errata Reported] =
RFC4028=20
    (1681)<BR>&gt;<BR>&gt; On Mon, 2009-02-09 at 13:03 -0600, Robert =
Sparks=20
    wrote:<BR>&gt; &gt; I agree - the example is not flawed (at least in =
the way=20
    the errata<BR>&gt; &gt; reports).<BR>&gt; &gt;<BR>&gt; &gt; Muthu =
seems to=20
    troubled by the reuse of the call-id and from tag when<BR>&gt; &gt; =
the=20
    initial transaction didn't create a dialog.<BR>&gt; &gt; While doing =
so is=20
    not required by the specification, nothing makes it<BR>&gt; &gt; =
illegal=20
    either.<BR>&gt;<BR>&gt; Are we agreed that "doing so is not required =
by the=20
    specification"? &nbsp;In<BR>&gt; RFC 3261, requests that are retried =
after=20
    401, 407, 413, 415, 416, and<BR>&gt; 420 responses are required to =
have the=20
    same from-tag as the original<BR>&gt; request. &nbsp;Is 422=20
    different?<BR>&gt;<BR>&gt; Dale<BR>&gt;<BR>&gt;<BR>&gt;=20
    _______________________________________________<BR>&gt; Sip mailing =
list=20
    &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>&gt; =
This=20
    list is for NEW development of the core SIP Protocol<BR>&gt; Use <A=20
    =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
    for questions on current sip<BR>&gt; Use <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
    the application of=20
    sip<BR>_______________________________________________<BR>Sip =
mailing list=20
    &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
    =
target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This =
list is=20
    for NEW development of the core SIP Protocol<BR>Use <A=20
    =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</A>=20
    for questions on current sip<BR>Use <A=20
    href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on=20
    the application of=20
sip<BR></DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></BODY></HTML>

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References: <200902091736.n19HaYee027174@boreas.isi.edu> <9B2A061A1137254BBE4F4B2CD843646A0FD3B1DE5F@mbx02.citservers.local> <A560AE7E-8E26-4D6D-B487-B98A3EDAB964@nostrum.com> <1234214049.4033.62.camel@victoria-pingtel-com.us.nortel.com> <9B2A061A1137254BBE4F4B2CD843646A0FD3B1DF1E@mbx02.citservers.local> <8c400f2e0902092057y5b98ce09nc69e5748d6032faa@mail.gmail.com> <C7FFFFDD779F2047A0FBAC811C5C5A0007B4B824@xmb-rtp-201.amer.cisco.com>
Date: Tue, 10 Feb 2009 20:25:43 +0530
Message-ID: <8c400f2e0902100655g3e62afafmec4e5f90a5fa950f@mail.gmail.com>
From: Muthu Arul <muthu.arul@gmail.com>
To: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
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Cc: SIP <sip@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Another source of confusion is that section 8.1.3.5 of RFC 3261 says the new
INVITE request SHOULD use same value of Call-ID, To and From as the previous
INVITE request. It is not clear whether it means just the same value of the
URI portion of To and From or also the same tags.

If it means the same tags then it is probably wrong --the new INVITE request
can not have a To tag, since the dialog is already gone.

Muthu

On Tue, Feb 10, 2009 at 1:44 PM, Sanjay Sinha (sanjsinh) <sanjsinh@cisco.com
> wrote:

>  Pl. see inline ...
>
>  ------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of *Muthu
> Arul
> *Sent:* Tuesday, February 10, 2009 10:27 AM
> *To:* Brett Tate
> *Cc:* SIP
> *Subject:* Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>
>  If the From tag doesn't change, can the UAS retrain the same To tag as
> well?
> [Sanjay Sinha (sanjsinh)]  When can UAS retain to-tag, when it is rejecting
> *resubmitted* request?
>
>  If the new INVITE reaches the UAS before the ACK for the previous INVITE,
> what should the UAS do?
> [Sanjay Sinha (sanjsinh)] Since the resubmitted request will not have a
> to-tag, even if Call-Id and from-tag is same, it wont match any ongoing
> transaction and so should be treated as a new request.
>
> Should it reject the new INVITE and expect the UAC to retry again?
>
> I thought it would have been much simpler had the From/To tag changed b/w
> the INVITE messages..
> [Sanjay Sinha (sanjsinh)]  It would have been better, but it is not illegal
> for Call-id + from-tag to not change. Besides, it is an example and should
> be treated as such..
>
> thanks,
> Muthu
>
> On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <brett@broadsoft.com> wrote:
>
>> Based upon the following snippet, my understanding is that RFC 4028 is
>> correctly following the SHOULD.
>>
>> RFC 3261 section 8.1.3.5 snippet:
>>
>> "In all of the above cases, the request is retried by creating a new
>> request with the appropriate modifications.  This new request constitutes a
>> new transaction and SHOULD have the same value of the Call-ID, To, and From
>> of the previous request, but the CSeq should contain a new sequence number
>> that is one higher than the previous. With other 4xx responses, including
>> those yet to be defined, a retry may or may not be possible depending on the
>> method and the use case."
>>
>> > -----Original Message-----
>> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Dale
>> > Worley
>> > Sent: Monday, February 09, 2009 4:14 PM
>> > To: SIP
>> > Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>> >
>> > On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:
>> > > I agree - the example is not flawed (at least in the way the errata
>> > > reports).
>> > >
>> > > Muthu seems to troubled by the reuse of the call-id and from tag when
>> > > the initial transaction didn't create a dialog.
>> > > While doing so is not required by the specification, nothing makes it
>> > > illegal either.
>> >
>> > Are we agreed that "doing so is not required by the specification"?  In
>> > RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and
>> > 420 responses are required to have the same from-tag as the original
>> > request.  Is 422 different?
>> >
>> > Dale
>> >
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www.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://www.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
>>
>
>

--0016e64f809074d30f046291ae0d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Another source of confusion is that section 8.1.3.5 of RFC 3261 says the ne=
w INVITE request SHOULD use same value of Call-ID, To and From as the previ=
ous INVITE request. It is not clear whether it means just the same value of=
 the URI portion of To and From or also the same tags.<br>
<br>If it means the same tags then it is probably wrong --the new INVITE re=
quest can not have a To tag, since the dialog is already gone.<br><br>Muthu=
<br><br><div class=3D"gmail_quote">On Tue, Feb 10, 2009 at 1:44 PM, Sanjay =
Sinha (sanjsinh) <span dir=3D"ltr">&lt;<a href=3D"mailto:sanjsinh@cisco.com=
">sanjsinh@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">



<div>
<div dir=3D"ltr" align=3D"left"><span></span><font face=3D"Arial"><font col=
or=3D"#0000ff"><font size=3D"2">P<span>l.=20
see inline ...</span></font></font></font><br></div>
<blockquote style=3D"border-left: 2px solid rgb(0, 0, 255); padding-left: 5=
px; margin-left: 5px; margin-right: 0px;">
  <div dir=3D"ltr" align=3D"left" lang=3D"en-us">
  <hr>
  <font size=3D"2" face=3D"Tahoma"><b>From:</b> <a href=3D"mailto:sip-bounc=
es@ietf.org" target=3D"_blank">sip-bounces@ietf.org</a>=20
  [mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bou=
nces@ietf.org</a>] <b>On Behalf Of </b>Muthu Arul<br><b>Sent:</b>=20
  Tuesday, February 10, 2009 10:27 AM<br><b>To:</b> Brett Tate<br><b>Cc:</b=
>=20
  SIP<div class=3D"Ih2E3d"><br><b>Subject:</b> Re: [Sip] [Technical Errata =
Reported] RFC4028=20
  (1681)<br></div></font><br></div>
  <div></div>
  <div><div class=3D"Ih2E3d">If the From tag doesn&#39;t change, can the UA=
S retrain the same To tag as=20
  well? <br></div><span><font color=3D"#0000ff" size=3D"2" face=3D"Arial">[=
Sanjay Sinha (sanjsinh)]&nbsp;&nbsp;When&nbsp;can UAS retain to-tag,=20
  when it is rejecting *resubmitted* request?</font></span></div>
  <div><span></span>&nbsp;</div>
  <div><div class=3D"Ih2E3d"><span></span><span>&nbsp;</span>If the new INV=
ITE reaches the UAS before=20
  the ACK for the previous INVITE, what should the UAS do? <br></div><span>=
<font color=3D"#0000ff" size=3D"2" face=3D"Arial">[Sanjay Sinha=20
  (sanjsinh)]&nbsp;Since the resubmitted request will not have a to-tag,=20
  even&nbsp;if Call-Id and from-tag is same,&nbsp;it wont match any ongoing=
=20
  transaction and so should be treated as a new=20
  request.&nbsp;</font></span></div>
  <div><span><font color=3D"#0000ff" size=3D"2" face=3D"Arial"></font></spa=
n>&nbsp;</div>
  <div><div class=3D"Ih2E3d">Should it reject the new INVITE and expect the=
 UAC to retry=20
  again?<br><br>I thought it would have been much simpler had the From/To t=
ag=20
  changed b/w the INVITE messages..<br></div><span><font color=3D"#0000ff" =
size=3D"2" face=3D"Arial">[Sanjay Sinha (sanjsinh)]&nbsp; It&nbsp;would=20
  have&nbsp;been better, but it is not illegal for Call-id + from-tag to=20
  not&nbsp;change. Besides, it is an example and should be treated as=20
  such..&nbsp;</font></span><br><br>thanks,<br>Muthu<br><br></div><div><div=
></div><div class=3D"Wj3C7c">
  <div class=3D"gmail_quote">On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:brett@broadsoft.com" target=3D"_blank=
">brett@broadsoft.com</a>&gt;</span>=20
wrote:<br>
  <blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204=
, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Based=20
    upon the following snippet, my understanding is that RFC 4028 is correc=
tly=20
    following the SHOULD.<br><br>RFC 3261 section 8.1.3.5 snippet:<br><br>&=
quot;In=20
    all of the above cases, the request is retried by creating a new reques=
t=20
    with the appropriate modifications. &nbsp;This new request constitutes =
a new=20
    transaction and SHOULD have the same value of the Call-ID, To, and From=
 of=20
    the previous request, but the CSeq should contain a new sequence number=
 that=20
    is one higher than the previous. With other 4xx responses, including th=
ose=20
    yet to be defined, a retry may or may not be possible depending on the=
=20
    method and the use case.&quot;<br>
    <div>
    <div></div>
    <div><br>&gt; -----Original Message-----<br>&gt; From: <a href=3D"mailt=
o:sip-bounces@ietf.org" target=3D"_blank">sip-bounces@ietf.org</a> [mailto:=
<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounces@ietf.=
org</a>] On Behalf Of=20
    Dale<br>&gt; Worley<br>&gt; Sent: Monday, February 09, 2009 4:14 PM<br>=
&gt;=20
    To: SIP<br>&gt; Subject: Re: [Sip] [Technical Errata Reported] RFC4028=
=20
    (1681)<br>&gt;<br>&gt; On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks=
=20
    wrote:<br>&gt; &gt; I agree - the example is not flawed (at least in th=
e way=20
    the errata<br>&gt; &gt; reports).<br>&gt; &gt;<br>&gt; &gt; Muthu seems=
 to=20
    troubled by the reuse of the call-id and from tag when<br>&gt; &gt; the=
=20
    initial transaction didn&#39;t create a dialog.<br>&gt; &gt; While doin=
g so is=20
    not required by the specification, nothing makes it<br>&gt; &gt; illega=
l=20
    either.<br>&gt;<br>&gt; Are we agreed that &quot;doing so is not requir=
ed by the=20
    specification&quot;? &nbsp;In<br>&gt; RFC 3261, requests that are retri=
ed after=20
    401, 407, 413, 415, 416, and<br>&gt; 420 responses are required to have=
 the=20
    same from-tag as the original<br>&gt; request. &nbsp;Is 422=20
    different?<br>&gt;<br>&gt; Dale<br>&gt;<br>&gt;<br>&gt;=20
    _______________________________________________<br>&gt; Sip mailing lis=
t=20
    &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/sip</a><br>&gt; This=20
    list is for NEW development of the core SIP Protocol<br>&gt; Use <a hre=
f=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank">sip-impleme=
ntors@cs.columbia.edu</a>=20
    for questions on current sip<br>&gt; Use <a href=3D"mailto:sipping@ietf=
.org" target=3D"_blank">sipping@ietf.org</a> for new developments on=20
    the application of=20
    sip<br>_______________________________________________<br>Sip mailing l=
ist=20
    &nbsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/sip</a><br>This list is=20
    for NEW development of the core SIP Protocol<br>Use <a href=3D"mailto:s=
ip-implementors@cs.columbia.edu" target=3D"_blank">sip-implementors@cs.colu=
mbia.edu</a>=20
    for questions on current sip<br>Use <a href=3D"mailto:sipping@ietf.org"=
 target=3D"_blank">sipping@ietf.org</a> for new developments on=20
    the application of=20
sip<br></div></div></blockquote></div><br></div></div></blockquote></div>
</blockquote></div><br>

--0016e64f809074d30f046291ae0d--

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From: Bob Penfield <BPenfield@acmepacket.com>
To: Muthu Arul <muthu.arul@gmail.com>, "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
Date: Tue, 10 Feb 2009 11:56:47 -0500
Thread-Topic: [Sip] [Technical Errata Reported] RFC4028 (1681)
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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The text in 8.1.3.5 means that the From and To headers in the new INVITE sh=
ould have the same value including tags as the rejected INVITE. If it was a=
n initial (out-of-dialog) INVITE, there will be no to-tag in the To, and th=
us the re-submitted request will not have a to-tag either. There is no need=
 to change the Call-ID or the from tag because no dialogs were created.

If it is an in-dialog request, it must have the same tags otherwise it will=
 not match the dialog. This could happen if authentication is required for =
in-dialog requests and the UAC did not include the credentials, or the cred=
entials are stale.

>From an implementation point of view, it is simpler if the procedures for r=
e-submitting requests for repairable 4xx errors are the same for in-dialog =
and out-of-dialog requests.

I see no reason to change the Call-ID or from tag.

As for the question "If the From tag doesn't change, can the UAS retrain th=
e same To tag as well?", I would say that UAS should not use the same to-ta=
g for a new request.

cheers,
(-:bob

________________________________
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Muthu=
 Arul
Sent: Tuesday, February 10, 2009 9:56 AM
To: Sanjay Sinha (sanjsinh)
Cc: SIP; Brett Tate
Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)

Another source of confusion is that section 8.1.3.5 of RFC 3261 says the ne=
w INVITE request SHOULD use same value of Call-ID, To and From as the previ=
ous INVITE request. It is not clear whether it means just the same value of=
 the URI portion of To and From or also the same tags.

If it means the same tags then it is probably wrong --the new INVITE reques=
t can not have a To tag, since the dialog is already gone.

Muthu

On Tue, Feb 10, 2009 at 1:44 PM, Sanjay Sinha (sanjsinh) <sanjsinh@cisco.co=
m<mailto:sanjsinh@cisco.com>> wrote:
Pl. see inline ...
________________________________
From: sip-bounces@ietf.org<mailto:sip-bounces@ietf.org> [mailto:sip-bounces=
@ietf.org<mailto:sip-bounces@ietf.org>] On Behalf Of Muthu Arul
Sent: Tuesday, February 10, 2009 10:27 AM
To: Brett Tate
Cc: SIP

Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)

If the From tag doesn't change, can the UAS retrain the same To tag as well=
?
[Sanjay Sinha (sanjsinh)]  When can UAS retain to-tag, when it is rejecting=
 *resubmitted* request?

 If the new INVITE reaches the UAS before the ACK for the previous INVITE, =
what should the UAS do?
[Sanjay Sinha (sanjsinh)] Since the resubmitted request will not have a to-=
tag, even if Call-Id and from-tag is same, it wont match any ongoing transa=
ction and so should be treated as a new request.

Should it reject the new INVITE and expect the UAC to retry again?

I thought it would have been much simpler had the From/To tag changed b/w t=
he INVITE messages..
[Sanjay Sinha (sanjsinh)]  It would have been better, but it is not illegal=
 for Call-id + from-tag to not change. Besides, it is an example and should=
 be treated as such..

thanks,
Muthu

On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <brett@broadsoft.com<mailto:bre=
tt@broadsoft.com>> wrote:
Based upon the following snippet, my understanding is that RFC 4028 is corr=
ectly following the SHOULD.

RFC 3261 section 8.1.3.5 snippet:

"In all of the above cases, the request is retried by creating a new reques=
t with the appropriate modifications.  This new request constitutes a new t=
ransaction and SHOULD have the same value of the Call-ID, To, and From of t=
he previous request, but the CSeq should contain a new sequence number that=
 is one higher than the previous. With other 4xx responses, including those=
 yet to be defined, a retry may or may not be possible depending on the met=
hod and the use case."

> -----Original Message-----
> From: sip-bounces@ietf.org<mailto:sip-bounces@ietf.org> [mailto:sip-bounc=
es@ietf.org<mailto:sip-bounces@ietf.org>] On Behalf Of Dale
> Worley
> Sent: Monday, February 09, 2009 4:14 PM
> To: SIP
> Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
>
> On Mon, 2009-02-09 at 13:03 -0600, Robert Sparks wrote:
> > I agree - the example is not flawed (at least in the way the errata
> > reports).
> >
> > Muthu seems to troubled by the reuse of the call-id and from tag when
> > the initial transaction didn't create a dialog.
> > While doing so is not required by the specification, nothing makes it
> > illegal either.
>
> Are we agreed that "doing so is not required by the specification"?  In
> RFC 3261, requests that are retried after 401, 407, 413, 415, 416, and
> 420 responses are required to have the same from-tag as the original
> request.  Is 422 different?
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.=
edu> for questions on current sip
> Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the=
 application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu<mailto:sip-implementors@cs.columbia.ed=
u> for questions on current sip
Use sipping@ietf.org<mailto:sipping@ietf.org> for new developments on the a=
pplication of sip



--_000_E6C2E8958BA59A4FB960963D475F7AC313DDD5BE6Dmail_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.6000.16788" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>The text in 8.1.3.5&nbsp;means that&nbsp;the From =
and To=20
headers in the new INVITE should have the same value including tags as=20
the&nbsp;rejected INVITE. If it was an initial (out-of-dialog) INVITE, ther=
e=20
will be no to-tag in the To, and thus the re-submitted request will not hav=
e a=20
to-tag either. There is no need to change the Call-ID or the from tag becau=
se no=20
dialogs were created.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>If it is an in-dialog request, it must have the sa=
me tags=20
otherwise it will not match the dialog. This could happen if authentication=
 is=20
required for in-dialog requests and the UAC did not include the credentials=
, or=20
the credentials are stale.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>From an implementation point of view, it is simple=
r if the=20
procedures for re-submitting requests for repairable 4xx errors&nbsp;are th=
e=20
same for in-dialog and out-of-dialog requests.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>I see no reason to change the Call-ID or from=20
tag.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>As for the question "If the From tag doesn't chang=
e, can=20
the UAS retrain the same To tag as well?", I would say that UAS should not =
use=20
the same to-tag for a new request.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>cheers,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D905554315-10022009><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>(-:bob</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Muthu Arul<BR><B>Sent:</B=
>=20
Tuesday, February 10, 2009 9:56 AM<BR><B>To:</B> Sanjay Sinha=20
(sanjsinh)<BR><B>Cc:</B> SIP; Brett Tate<BR><B>Subject:</B> Re: [Sip] [Tech=
nical=20
Errata Reported] RFC4028 (1681)<BR></FONT><BR></DIV>
<DIV></DIV>Another source of confusion is that section 8.1.3.5 of RFC 3261 =
says=20
the new INVITE request SHOULD use same value of Call-ID, To and From as the=
=20
previous INVITE request. It is not clear whether it means just the same val=
ue of=20
the URI portion of To and From or also the same tags.<BR><BR>If it means th=
e=20
same tags then it is probably wrong --the new INVITE request can not have a=
 To=20
tag, since the dialog is already gone.<BR><BR>Muthu<BR><BR>
<DIV class=3Dgmail_quote>On Tue, Feb 10, 2009 at 1:44 PM, Sanjay Sinha (san=
jsinh)=20
<SPAN dir=3Dltr>&lt;<A=20
href=3D"mailto:sanjsinh@cisco.com">sanjsinh@cisco.com</A>&gt;</SPAN> wrote:=
<BR>
<BLOCKQUOTE class=3Dgmail_quote=20
style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204=
,204,204) 1px solid">
  <DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN></SPAN><FONT face=3DArial><FONT=20
  color=3D#0000ff><FONT size=3D2>P<SPAN>l. see inline=20
  ...</SPAN></FONT></FONT></FONT><BR></DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(0,0,255) 2=
px solid; MARGIN-RIGHT: 0px">
    <DIV lang=3Den-us dir=3Dltr align=3Dleft>
    <HR>
    <FONT face=3DTahoma size=3D2><B>From:</B> <A href=3D"mailto:sip-bounces=
@ietf.org"=20
    target=3D_blank>sip-bounces@ietf.org</A> [mailto:<A=20
    href=3D"mailto:sip-bounces@ietf.org" target=3D_blank>sip-bounces@ietf.o=
rg</A>]=20
    <B>On Behalf Of </B>Muthu Arul<BR><B>Sent:</B> Tuesday, February 10, 20=
09=20
    10:27 AM<BR><B>To:</B> Brett Tate<BR><B>Cc:</B> SIP
    <DIV class=3DIh2E3d><BR><B>Subject:</B> Re: [Sip] [Technical Errata Rep=
orted]=20
    RFC4028 (1681)<BR></DIV></FONT><BR></DIV>
    <DIV></DIV>
    <DIV>
    <DIV class=3DIh2E3d>If the From tag doesn't change, can the UAS retrain=
 the=20
    same To tag as well? <BR></DIV><SPAN><FONT face=3DArial color=3D#0000ff=
=20
    size=3D2>[Sanjay Sinha (sanjsinh)]&nbsp;&nbsp;When&nbsp;can UAS retain =
to-tag,=20
    when it is rejecting *resubmitted* request?</FONT></SPAN></DIV>
    <DIV><SPAN></SPAN>&nbsp;</DIV>
    <DIV>
    <DIV class=3DIh2E3d><SPAN></SPAN><SPAN>&nbsp;</SPAN>If the new INVITE r=
eaches=20
    the UAS before the ACK for the previous INVITE, what should the UAS do?=
=20
    <BR></DIV><SPAN><FONT face=3DArial color=3D#0000ff size=3D2>[Sanjay Sin=
ha=20
    (sanjsinh)]&nbsp;Since the resubmitted request will not have a to-tag,=
=20
    even&nbsp;if Call-Id and from-tag is same,&nbsp;it wont match any ongoi=
ng=20
    transaction and so should be treated as a new=20
    request.&nbsp;</FONT></SPAN></DIV>
    <DIV><SPAN><FONT face=3DArial color=3D#0000ff size=3D2></FONT></SPAN>&n=
bsp;</DIV>
    <DIV>
    <DIV class=3DIh2E3d>Should it reject the new INVITE and expect the UAC =
to=20
    retry again?<BR><BR>I thought it would have been much simpler had the=20
    From/To tag changed b/w the INVITE messages..<BR></DIV><SPAN><FONT=20
    face=3DArial color=3D#0000ff size=3D2>[Sanjay Sinha (sanjsinh)]&nbsp;=20
    It&nbsp;would have&nbsp;been better, but it is not illegal for Call-id =
+=20
    from-tag to not&nbsp;change. Besides, it is an example and should be tr=
eated=20
    as such..&nbsp;</FONT></SPAN><BR><BR>thanks,<BR>Muthu<BR><BR></DIV>
    <DIV>
    <DIV></DIV>
    <DIV class=3DWj3C7c>
    <DIV class=3Dgmail_quote>On Tue, Feb 10, 2009 at 2:55 AM, Brett Tate <S=
PAN=20
    dir=3Dltr>&lt;<A href=3D"mailto:brett@broadsoft.com"=20
    target=3D_blank>brett@broadsoft.com</A>&gt;</SPAN> wrote:<BR>
    <BLOCKQUOTE class=3Dgmail_quote=20
    style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb=
(204,204,204) 1px solid">Based=20
      upon the following snippet, my understanding is that RFC 4028 is corr=
ectly=20
      following the SHOULD.<BR><BR>RFC 3261 section 8.1.3.5 snippet:<BR><BR=
>"In=20
      all of the above cases, the request is retried by creating a new requ=
est=20
      with the appropriate modifications. &nbsp;This new request constitute=
s a=20
      new transaction and SHOULD have the same value of the Call-ID, To, an=
d=20
      From of the previous request, but the CSeq should contain a new seque=
nce=20
      number that is one higher than the previous. With other 4xx responses=
,=20
      including those yet to be defined, a retry may or may not be possible=
=20
      depending on the method and the use case."<BR>
      <DIV>
      <DIV></DIV>
      <DIV><BR>&gt; -----Original Message-----<BR>&gt; From: <A=20
      href=3D"mailto:sip-bounces@ietf.org" target=3D_blank>sip-bounces@ietf=
.org</A>=20
      [mailto:<A href=3D"mailto:sip-bounces@ietf.org"=20
      target=3D_blank>sip-bounces@ietf.org</A>] On Behalf Of Dale<BR>&gt;=20
      Worley<BR>&gt; Sent: Monday, February 09, 2009 4:14 PM<BR>&gt; To:=20
      SIP<BR>&gt; Subject: Re: [Sip] [Technical Errata Reported] RFC4028=20
      (1681)<BR>&gt;<BR>&gt; On Mon, 2009-02-09 at 13:03 -0600, Robert Spar=
ks=20
      wrote:<BR>&gt; &gt; I agree - the example is not flawed (at least in =
the=20
      way the errata<BR>&gt; &gt; reports).<BR>&gt; &gt;<BR>&gt; &gt; Muthu=
=20
      seems to troubled by the reuse of the call-id and from tag when<BR>&g=
t;=20
      &gt; the initial transaction didn't create a dialog.<BR>&gt; &gt; Whi=
le=20
      doing so is not required by the specification, nothing makes it<BR>&g=
t;=20
      &gt; illegal either.<BR>&gt;<BR>&gt; Are we agreed that "doing so is =
not=20
      required by the specification"? &nbsp;In<BR>&gt; RFC 3261, requests t=
hat=20
      are retried after 401, 407, 413, 415, 416, and<BR>&gt; 420 responses =
are=20
      required to have the same from-tag as the original<BR>&gt; request.=20
      &nbsp;Is 422 different?<BR>&gt;<BR>&gt; Dale<BR>&gt;<BR>&gt;<BR>&gt;=
=20
      _______________________________________________<BR>&gt; Sip mailing l=
ist=20
      &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
      target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>&gt;=
 This=20
      list is for NEW development of the core SIP Protocol<BR>&gt; Use <A=20
      href=3D"mailto:sip-implementors@cs.columbia.edu"=20
      target=3D_blank>sip-implementors@cs.columbia.edu</A> for questions on=
=20
      current sip<BR>&gt; Use <A href=3D"mailto:sipping@ietf.org"=20
      target=3D_blank>sipping@ietf.org</A> for new developments on the appl=
ication=20
      of sip<BR>_______________________________________________<BR>Sip mail=
ing=20
      list &nbsp;<A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
      target=3D_blank>https://www.ietf.org/mailman/listinfo/sip</A><BR>This=
 list=20
      is for NEW development of the core SIP Protocol<BR>Use <A=20
      href=3D"mailto:sip-implementors@cs.columbia.edu"=20
      target=3D_blank>sip-implementors@cs.columbia.edu</A> for questions on=
=20
      current sip<BR>Use <A href=3D"mailto:sipping@ietf.org"=20
      target=3D_blank>sipping@ietf.org</A> for new developments on the appl=
ication=20
      of=20
sip<BR></DIV></DIV></BLOCKQUOTE></DIV><BR></DIV></DIV></BLOCKQUOTE></DIV></=
BLOCKQUOTE></DIV><BR></BODY></HTML>

--_000_E6C2E8958BA59A4FB960963D475F7AC313DDD5BE6Dmail_--

From dworley@nortel.com  Tue Feb 10 09:12:31 2009
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From: "Dale Worley" <dworley@nortel.com>
To: Bob Penfield <BPenfield@acmepacket.com>
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Cc: "Sanjay Sinha \(sanjsinh\)" <sanjsinh@cisco.com>, SIP <sip@ietf.org>, Brett Tate <brett@broadsoft.com>
Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1681)
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On Tue, 2009-02-10 at 11:56 -0500, Bob Penfield wrote:
> The text in 8.1.3.5 means that the From and To headers in the new
> INVITE should have the same value including tags as the rejected
> INVITE. If it was an initial (out-of-dialog) INVITE, there will be no
> to-tag in the To, and thus the re-submitted request will not have a
> to-tag either. There is no need to change the Call-ID or the from tag
> because no dialogs were created.
>  
> If it is an in-dialog request, it must have the same tags otherwise it
> will not match the dialog. This could happen if authentication is
> required for in-dialog requests and the UAC did not include the
> credentials, or the credentials are stale.
>  
> From an implementation point of view, it is simpler if the procedures
> for re-submitting requests for repairable 4xx errors are the same for
> in-dialog and out-of-dialog requests.
>  
> I see no reason to change the Call-ID or from tag.

All of the above matches my understanding and the discussions that I've
seen here.
 
> As for the question "If the From tag doesn't change, can the UAS
> retrain the same To tag as well?", I would say that UAS should not use
> the same to-tag for a new request.

I'm sure that the UAS should not use the same to-tag in its response to
the new request -- the first request did not establish a dialog, so the
second request could not be part of that dialog.  And indeed, it would
be difficult to build a UAS that did use the same to-tag, as it would
have to remember the to-tag it used in the first request and use it in
the second -- but nonetheless, if it received two copies of the first
INVITE and rejected both, it would have to use different to-tags for the
two error responses.

Dale



From web-usrn@ISI.EDU  Tue Feb 10 13:34:54 2009
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Subject: [Sip] [Technical Errata Reported] RFC3261 (1682)
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The following errata report has been submitted for RFC3261,
"SIP: Session Initiation Protocol".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3261&eid=1682

--------------------------------------
Type: Technical
Reported by: Iñaki Baz Castillo <ibc@aliax.net>

Section: 20.7

Original Text
-------------
Authorization: Digest username="Alice", realm="atlanta.com",
  nonce="84a4cc6f3082121f32b42a2187831a9e",
  response="7587245234b3434cc3412213e5f113a5432"

Corrected Text
--------------
Authorization: Digest username="Alice", realm="atlanta.com",
  nonce="84a4cc6f3082121f32b42a2187831a9e",
  response="7587245234b3434cc3412213e5f113a5"

Notes
-----
'response' field in original example has 35 hexadecimal characters while they must be 32:

  dresponse         =  "response" EQUAL request-digest
  request-digest    =  LDQUOT 32LHEX RDQUOT

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3261 (draft-ietf-sip-rfc2543bis-09)
--------------------------------------
Title               : SIP: Session Initiation Protocol
Publication Date    : June 2002
Author(s)           : J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From mary.barnes@nortel.com  Wed Feb 11 08:42:35 2009
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Dale Worley" <dworley@nortel.com>, "SIP" <sip@ietf.org>
Subject: Re: [Sip] Problems with History-Info
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Hi Dale,

Sorry for the delay in responding - responses below [MB].

Mary.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Worley, Dale (BL60:9D30)
Sent: Sunday, February 01, 2009 1:54 PM
To: SIP
Subject: [Sip] Problems with History-Info

To get discussion going for 2009, let me start with some complaints
about the History-Info header.

One problem is identifying which value in the History-Info header
corresponds to the request carrying the header, that is, which branch of
the History-Info request tree represents the current request.

Reading RFC 4244, it's clear that what is intended is that the
History-Info values will be written in depth-first tree-walk order, and
that syntactically *last* value corresponds to the request carrying the
History-Info header.  Neither of these principles is stated explicitly
as far as I can tell.=20

[MB] RFC 4244 is clear that the entries are in a tree structure - you
are correct that we don't identify it as a "depth-first" and perhaps we
can add that, although I think it's quite clear from examples and
normative text.  As far as  your concern wrt to "synctactically *last*
value corresponds to the request carrying HI", I'm not sure what you
mean here. [/MB]

Difficulty arises when there is parallel forking, or any similar
situation when a request has multiple redirects or retargets that become
known in one processing step.  If one maintains the rule that the
enclosing request corresponds to the last History-Info value, one must
choose among:

[MB] There is not a problem here. The spec is very clear about what
happens when a proxy (sip:1@example.com) retargets either sequentially
or in parallel.  And, RFC 4244 defines a 4th choice not listed below: =20

      INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1B@example.com>;index=3D1.2

This is specified in section 4.3.3.1.3, in particular steps 2 and 3
result in the above HI entries with an example of this behavior in
section 4.5 (sorry no figure number - something we should add in the
-bis).  Per the spec, there are never any entries that are identical -
if there, it's a bug. If you think there is text in RFC 4244 that
contradicts this, please identify it and we can clarify it in the -bis.=20
[/MB]


(Assume that sip:1@example.com has been retargeted to sip:1A@example.com
and sip:1B@example.com.)=20

1.each derived request carries only one of the new History-Info values

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1B@example.com>;index=3D1.1

2.each derived request carries only the new History-Info values that
  are "before" (in some ordering) the one for the current request

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1,
                      <sip:1B@example.com>;index=3D1.2

3.each derived request carries all of the new History-Info values, but
  in each request, they are ordered so that the value corresponding to
  the request is last

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1B@example.com>;index=3D1.1,
                      <sip:1A@example.com>;index=3D1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1,
                      <sip:1B@example.com>;index=3D1.2

Solutions 1 and 3 are troublesome because the generated History-Info
values are not consistent between the derived requests, and solutions
1 and 2 are troublesome because the downstream network elements are
given less information than is available about the forking of the
request.

Note that this isn't just a theoretical problem,
draft-rosenberg-sip-target-uri-delivery-00 is already running up against
it (section 5):  "Furthermore, if that URI had an 'index' of N, the URIs
with indices N.M for all M, are the registered contacts to that URI."
If the I-D is taken at face value, solution 3 must be used, giving
different History-Info values on different derived requests.

Let us then consider putting all known targets of a request in
History-Info, but not require that the last History-Info value
corresponds to the containing request.  In terms of the above example,
both derived requests would have the same History-Info header:

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1,
                      <sip:1B@example.com>;index=3D1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1,
                      <sip:1B@example.com>;index=3D1.2

If we do this, we need to be able to identify the value that corresponds
to the current request by some method other than the index values and
syntactic ordering.

In a perfect world, we could compare the request-URI with the URIs in
History-Info to see which one matches.  However, this fails if:

- two URIs in History-Info are the same (In theory, this should not
  happen, but it can happen sensibly if two derived requests differed
  substantially in some manner which could affect the ultimate
  resolution of the request-URI; e.g., caller preferences.)

- if an intermediate proxy has changed the request-URI but not updated
  History-Info (In this case, we want to know which History-Info value
  corresponds to the most recent parent request that was logged in
  History-Info.)

A more reliable solution would be to add a "current" marker to the leaf
of the History-Info tree that corresponds to the containing request.  In
the example above:

	INVITE sip:1A@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1;current,
                      <sip:1B@example.com>;index=3D1.2

	INVITE sip:1B@example.com
	History-Info: <sip:1@example.com>;index=3D1,
                      <sip:1A@example.com>;index=3D1.1,
                      <sip:1B@example.com>;index=3D1.2;current

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

A second problem with History-Info as it is currently structured is in
the handling of requests generated as a consequence of recursing on 3xx
responses.

History-Info as it is now defined correctly shows the "forking
structure" of a request.  But we also want to know the "derivation" of
the current request-URI, the sequence of transformations from the
original request-URI (presumably the same as the To URI) to the current
request-URI.  Assuming that the transformations are done in the ordinary
manner by proxies, History-Info carries the needed information.  From
RFC 4244 section 4.5:

   UA1                      Proxy1                    Proxy2
UA2

   |                          |                         |
|
   |--INVITE sip:Bob@P1... -->|                         |
|
      History-Info: <sip:Bob@P1.example.com>;index=3D1
   |                          |-INVITE sip:Bob@P2... -->|
|
                                History-Info:
<sip:Bob@P1.example.com>;index=3D1,
=20
<sip:Bob@P2.example.com>;index=3D1.1
   |                          |                         |
|
   |                          |                         |-INVITE
sip:User2@UA2...-->|
                                                          History-Info:
<sip:Bob@P1.example.com>;index=3D1,
=20
<sip:Bob@P2.example.com>;index=3D1.1,
=20
<sip:User2@UA2.example.com>;index=3D1.1.1

In the last INVITE, starting with the last URI and working our way up
the tree, we see that the URI sequence is:

sip:Bob@P1.example.com -> sip:Bob@P2.example.com ->
sip:User2@UA2.example.com

However, if one of the transformations is supplied by a 3xx response
(e.g., if the request is sent to a "redirect server"), the History-Info
data does not effectively show the source of the current
request-URI:

   UA1                      Proxy1                    Proxy2      UA2

   |                          |                         |          |
   |--INVITE sip:Bob@P1... -->|                         |          |
      History-Info: <sip:Bob@P1.example.com>;index=3D1
   |                          |-INVITE sip:Bob@P2... -->|          |
                                History-Info:
<sip:Bob@P1.example.com>;index=3D1,
=20
<sip:Bob@P2.example.com>;index=3D1.1
   |                          |<-- 302
                                   Contact: <sip:User2@UA2.example.com>
   |                          |                         |          |
   |                          |-INVITE sip:User2@UA2...----------->|
                                History-Info:
<sip:Bob@P1.example.com>;index=3D1,
=20
<sip:Bob@P2.example.com?Reason=3DSIP%3Bcause%3D302>;index=3D1.1,
=20
<sip:User2@UA2.example.com>;index=3D1.2

The apparent URI sequence is:

sip:Bob@P1.example.com -> sip:User2@UA2.example.com

There is no unambiguous way determine that the target
sip:User2@UA2.example.com came from the 302 response to the request to
sip:Bob@P2.example.com. =20

[MB] I don't see a problem here, as you can indeed determine that the
request was retargetted to User2 as a result of the 3xx - Proxy 1
absolutely knows that and that's the reason why the "reason" is added to
the previous HI entry - i.e., the rule being that the reason is
associated with the "Why the request was retargeted?" from the previous
HI entry (Bob@P2.example.com) to the new HI entry User@UA2.example.com).
Thus, you can derive the sequence from the above -i.e., it's ALL there
in that last INVITE.  It does require that you look for the reason, but
you really don't need the redundant data. An old, old version of this
document actually did have redundant data in that the "old" and "new"
were captured in each entry. The current approach is that if the "old"
isn't captured, then the entity retargeting captures it along with the
"new" to ensure that you have the entire chain. If you can point to the
specific text that seems to introduce the gap you suggest, that would be
helpful and we can make sure it's clear in the -bis.=20
[/MB]

What is needed is a pointer from a History-Info value to the value
representing the request which yielded the 3xx response that cause the
value in question.  I suggest using a parameter named "3xx" for this
purpose, so that the last request in the above example becomes:

    INVITE sip:User2@UA2.example.com
    History-Info: <sip:Bob@P1.example.com>;index=3D1,
=09
<sip:Bob@P2.example.com?Reason=3DSIP%3Bcause%3D302>;index=3D1.1,
		  <sip:User2@UA2.example.com>;index=3D1.2;3xx=3D1.1

Which this additional information present, the predecessor of a listed
URI is either the URI of its 3xx value (if present), or its parent in
the tree.  This allows us to recover the URI transformation sequence:

sip:Bob@P1.example.com -> sip:Bob@P2.example.com ->
sip:User2@UA2.example.com

Dale


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From dworley@nortel.com  Wed Feb 11 13:44:27 2009
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To: "Mary Barnes" <mary.barnes@nortel.com>
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Subject: Re: [Sip] Problems with History-Info
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On Wed, 2009-02-11 at 11:44 -0500, Barnes, Mary (RICH2:AR00) wrote:
> > Reading RFC 4244, it's clear that what is intended is that the
> > History-Info values will be written in depth-first tree-walk order, and
> > that syntactically *last* value corresponds to the request carrying the
> > History-Info header.  Neither of these principles is stated explicitly
> > as far as I can tell. 
> 
> [MB] RFC 4244 is clear that the entries are in a tree structure - you
> are correct that we don't identify it as a "depth-first" and perhaps we
> can add that, although I think it's quite clear from examples and
> normative text.  As far as  your concern wrt to "synctactically *last*
> value corresponds to the request carrying HI", I'm not sure what you
> mean here. [/MB]

What I see is that all examples are presented in depth-first order, and
various operations are described in terms of "adding a value".  But I
don't see a specification that the values must be listed in depth-first
order, nor that the added values must go in a certain position in the
list of values.  Given that the values are tagged with indexes, I could
interpret the text to mean that the values are associated based on their
index values, and it was just convenience or accident that all examples
presented in the RFC were written in depth-first order.

In regard to "syntactically last", I was looking at the question of
knowing which entry in History-Info is meant to describe the request
that contains it.  It seems from the examples that the current request
is described by the linearly last (lowest-rightmost in tree order)
element, but it's not clear that is intended to be specified.  One could
search for the request-URI in the History-Info elements, but (1) URI
matching is treacherous, and (2) one can construct cases where the same
request-URI can sensibly appear in more than one incarnation of an
INVITE.  It seems that the intention is that the last element is
expected to match the current request, but I don't see that stated.

> Difficulty arises when there is parallel forking, or any similar
> situation when a request has multiple redirects or retargets that become
> known in one processing step.  If one maintains the rule that the
> enclosing request corresponds to the last History-Info value, one must
> choose among:
> 
> [MB] There is not a problem here. The spec is very clear about what
> happens when a proxy (sip:1@example.com) retargets either sequentially
> or in parallel.  And, RFC 4244 defines a 4th choice not listed below:  
> 
>       INVITE sip:1A@example.com
> 	History-Info: <sip:1@example.com>;index=1,
>                       <sip:1A@example.com>;index=1.1
> 
> 	INVITE sip:1B@example.com
> 	History-Info: <sip:1@example.com>;index=1,
>                       <sip:1B@example.com>;index=1.2
> 
> This is specified in section 4.3.3.1.3, in particular steps 2 and 3
> result in the above HI entries with an example of this behavior in
> section 4.5 (sorry no figure number - something we should add in the
> -bis).  Per the spec, there are never any entries that are identical -
> if there, it's a bug. If you think there is text in RFC 4244 that
> contradicts this, please identify it and we can clarify it in the -bis. 
> [/MB]

I'll grant I overlooked that example.  The implication is that the tree
presented in History-Info can be *incomplete*, there can be a 1.2 branch
without a 1.1 branch, not because the user used weird numbering, but
because there is a 1.1. branch that is not being described in this
History-Info -- something that is not so common in representations of
trees, and doesn't seem to be explicitly pointed out in the text.

In regard to 4.3.3.1.3, the text is not very clear if you don't already
know what it means:

        Retargeting within a Proxy - subsequent instance: For each
        subsequent retargeting of a request by the same proxy, another
        branch is added.  With the index for each new branch calculated
        by incrementing the last/lowest digit at the current level, the
        index in the next request forwarded by this same proxy,
        following the example above, would be 1.1.2.

What is not stated is that the first forked request gets the xxx.1 value
added to its History-Info, the second forked request gets the xxx.2
value added to its History-Info (but not the xxx.1 value), etc.  When I
read this, I assumed that "instance" meant History-Info value, and that
each forked request was expected to have the same set of History-Info
values, namely the whole set generated by the forking, of which the
first "instance" would be numbers xxx.1, etc.

> (Assume that sip:1@example.com has been retargeted to sip:1A@example.com
> and sip:1B@example.com.) 
> 
> 1.each derived request carries only one of the new History-Info values
> 
> 	INVITE sip:1A@example.com
> 	History-Info: <sip:1@example.com>;index=1,
>                       <sip:1A@example.com>;index=1.1
> 
> 	INVITE sip:1B@example.com
> 	History-Info: <sip:1@example.com>;index=1,
>                       <sip:1B@example.com>;index=1.1

So the correct answer is case 1, but modified in that the index for
sip:1B@example.com is 1.2; the numbering was globally consistent across
all the History-Info values, but that each value was a subset of the
global forking tree.

> [MB] I don't see a problem here, as you can indeed determine that the
> request was retargetted to User2 as a result of the 3xx - Proxy 1
> absolutely knows that and that's the reason why the "reason" is added to
> the previous HI entry - i.e., the rule being that the reason is
> associated with the "Why the request was retargeted?" from the previous
> HI entry (Bob@P2.example.com) to the new HI entry User@UA2.example.com).
> Thus, you can derive the sequence from the above -i.e., it's ALL there
> in that last INVITE.  It does require that you look for the reason, but
> you really don't need the redundant data. An old, old version of this
> document actually did have redundant data in that the "old" and "new"
> were captured in each entry. The current approach is that if the "old"
> isn't captured, then the entity retargeting captures it along with the
> "new" to ensure that you have the entire chain. If you can point to the
> specific text that seems to introduce the gap you suggest, that would be
> helpful and we can make sure it's clear in the -bis. 
> [/MB]

I was thinking of the case where there are two or more forks preceding
the current fork that have returned 3xx.  It seems that you'd get H-I
like:

History-Info:
    <sip:UserA@example.com>;index=1,
    <sip:UserA1@example.com?Reason=SIP;cause=302>;index=1.1,
    <sip:UserB@example.com?Reason=SIP;cause=302>;index=1.2,
    <sip:UserC@example.com>;index=1.3

>From that, I don't see how to determine whether UserC was generated from
UserA1 or UserB, or for that matter, if it was derived from UserA.

Perhaps there is an unstated rule that History-Info should contain only
that subset of the global forking tree that describes requests that are
logically ancestral to the request that contains it?

(Oh, yeah, and the example in RFC 4244 from which I derived the above
example is:

      F12 486 Busy Here Proxy1 -> UA1

      SIP/2.0  486 Busy Here
      Via: SIP/2.0/UDP example.net:5060
      From: Alice <sip:User1@example.net>
      To: Bob <sip:UserA@example.com>
      Call-Id: 12345600@example.net
      History-Info:  <sip:UserA@example.com>; index=1,
       <sip:UserA@ims.example.com?Reason=SIP;\
       cause=302; text="Moved Temporarily">;index=1.1,
       <sip:UserB@example.com?Reason=SIP;cause=480;\
       text="Temporarily Unavailable" >;index=1.2,
       <sip:UserC@example.com>;index=1.3
      CSeq: 1 INVITE
      Content-Length: 0

Note that it presents a case where one request-URI appears in more than
one History-Info value, which was a question I raised above.)

Dale



From brett@broadsoft.com  Wed Feb 11 16:00:08 2009
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From: Brett Tate <brett@broadsoft.com>
To: SIP <sip@ietf.org>
Date: Wed, 11 Feb 2009 16:00:09 -0800
Thread-Topic: rfc3261: default contact q-value
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Greetings,

Within RFC 3261, I was unable to find a default q-value for Contact.

Versions 2 through 4 of draft-ietf-sip-rfc2543bis indicated 0.5 as default.=
  It looks like it was unintentionally removed; however I'm not certain.

What is the default Contact q-value and where is it documented?

Thanks,
Brett


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From: "Kanumuri, Sreeram" <sreeramk@qualcomm.com>
To: Brett Tate <brett@broadsoft.com>, SIP <sip@ietf.org>
Date: Wed, 11 Feb 2009 16:11:38 -0800
Thread-Topic: rfc3261: default contact q-value
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Hi Brett,

The default value of q-value is 1.

I am not sure when but there was discussion in sip implementers list some t=
ime back on this.

It takes from HTTP where it is mentioned default is 1 when not specified.

HTH,
Sreeram

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Brett=
 Tate
Sent: Wednesday, February 11, 2009 4:00 PM
To: SIP
Subject: [Sip] rfc3261: default contact q-value

Greetings,

Within RFC 3261, I was unable to find a default q-value for Contact.

Versions 2 through 4 of draft-ietf-sip-rfc2543bis indicated 0.5 as default.=
  It looks like it was unintentionally removed; however I'm not certain.

What is the default Contact q-value and where is it documented?

Thanks,
Brett

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From frank.shearar@angband.za.org  Thu Feb 12 00:59:13 2009
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The second last paragraph of section refers to a
"active-signaling-active-attack-detect" category. Should that not be
"active-signalling-active-media-detect" ?

frank


From victor.pascual.avila@gmail.com  Thu Feb 12 02:00:43 2009
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Subject: Re: [Sip] rfc3261: default contact q-value
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On Thu, Feb 12, 2009 at 1:11 AM, Kanumuri, Sreeram
<sreeramk@qualcomm.com> wrote:
> The default value of q-value is 1.

https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-November/0112=
37.html

> It takes from HTTP where it is mentioned default is 1 when not specified.

https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-November/0112=
38.html

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

From brett@broadsoft.com  Thu Feb 12 05:42:12 2009
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Thanks for the responses, especially the reference to RFC 3841.

I guess the working group agreed to remove it from being "0.5" (or document=
ed within the rfc2543bis) when creating draft-ietf-sip-rfc2543bis-05 and ag=
reed to start documenting it as "1.0" within draft-ietf-sip-callerprefs-06.

Including the default value for "q" within the base specification sounds be=
tter to me; however documenting the default within RFC 3841 (or non SIP RFC=
s as some have indicated) might be adequate.


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Brett Tate
> Sent: Wednesday, February 11, 2009 7:00 PM
> To: SIP
> Subject: [Sip] rfc3261: default contact q-value
>=20
> Greetings,
>=20
> Within RFC 3261, I was unable to find a default q-value for Contact.
>=20
> Versions 2 through 4 of draft-ietf-sip-rfc2543bis indicated 0.5 as
> default.  It looks like it was unintentionally removed; however I'm not
> certain.
>=20
> What is the default Contact q-value and where is it documented?
>=20
> Thanks,
> Brett
>=20


From mary.barnes@nortel.com  Thu Feb 12 09:18:36 2009
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More responses inline below [MB2].

-----Original Message-----
From: Worley, Dale (BL60:9D30)=20
Sent: Wednesday, February 11, 2009 3:44 PM
To: Barnes, Mary (RICH2:AR00)
Cc: SIP
Subject: RE: [Sip] Problems with History-Info

On Wed, 2009-02-11 at 11:44 -0500, Barnes, Mary (RICH2:AR00) wrote:
> > Reading RFC 4244, it's clear that what is intended is that the=20
> > History-Info values will be written in depth-first tree-walk order,=20
> > and that syntactically *last* value corresponds to the request=20
> > carrying the History-Info header.  Neither of these principles is=20
> > stated explicitly as far as I can tell.
>=20
> [MB] RFC 4244 is clear that the entries are in a tree structure - you=20
> are correct that we don't identify it as a "depth-first" and perhaps=20
> we can add that, although I think it's quite clear from examples and=20
> normative text.  As far as  your concern wrt to "synctactically *last*

> value corresponds to the request carrying HI", I'm not sure what you=20
> mean here. [/MB]

What I see is that all examples are presented in depth-first order, and
various operations are described in terms of "adding a value".  But I
don't see a specification that the values must be listed in depth-first
order, nor that the added values must go in a certain position in the
list of values.  Given that the values are tagged with indexes, I could
interpret the text to mean that the values are associated based on their
index values, and it was just convenience or accident that all examples
presented in the RFC were written in depth-first order.

[MB2] With the indices, it actually doesn't matter if the entries are
added in depth first order. However, RFC 4244 does specify in section
4.3.3.1 that the entries MUST be added following any HI entry received
in the request.  Now, certainly, in the parallel forking situation, you
wouldn't know if an entry 1.2.1 was really added before an entry 1.3.1,
but that's more because the order of responses is entirely indeterminate
per core SIP when a request is forked. It might be sensible (but not
required for things to work) that the proxy that forked the requests
would keep the HI entries grouped logically. We could add a suggestion
to this effect in the -bis. [/MB2]=20

In regard to "syntactically last", I was looking at the question of
knowing which entry in History-Info is meant to describe the request
that contains it.  It seems from the examples that the current request
is described by the linearly last (lowest-rightmost in tree order)
element, but it's not clear that is intended to be specified.  One could
search for the request-URI in the History-Info elements, but (1) URI
matching is treacherous, and (2) one can construct cases where the same
request-URI can sensibly appear in more than one incarnation of an
INVITE.  It seems that the intention is that the last element is
expected to match the current request, but I don't see that stated.

[MB2] When a proxy receives a request that doesn't have an entry for the
hop that got the request to this proxy, then it is suggested to add an
entry, so if I understand you, yes, when a request reaches a proxy, the
last entry should be the same as the request URI in the request. I don't
know how that can be stated more clearly as that's a fundamental to how
HI is captured and comprises the basic definition of the
"Targeted-to-URI" field in the header per section 4.1.  It's also
described in section 4.3.1 (UAC behavior).  It may not be quite so clear
in section 4.3.3.1 (Proxy behavior) and we could be more explicit about
that when we discuss that "if the request doesn't contain a History-Info
header" to state the "hi-targeted-to-uri" is set to the URI in the
request". [/MB2]

> Difficulty arises when there is parallel forking, or any similar=20
> situation when a request has multiple redirects or retargets that=20
> become known in one processing step.  If one maintains the rule that=20
> the enclosing request corresponds to the last History-Info value, one=20
> must choose among:
>=20
> [MB] There is not a problem here. The spec is very clear about what=20
> happens when a proxy (sip:1@example.com) retargets either sequentially

> or in parallel.  And, RFC 4244 defines a 4th choice not listed below:
>=20
>       INVITE sip:1A@example.com
> 	History-Info: <sip:1@example.com>;index=3D1,
>                       <sip:1A@example.com>;index=3D1.1
>=20
> 	INVITE sip:1B@example.com
> 	History-Info: <sip:1@example.com>;index=3D1,
>                       <sip:1B@example.com>;index=3D1.2
>=20
> This is specified in section 4.3.3.1.3, in particular steps 2 and 3=20
> result in the above HI entries with an example of this behavior in=20
> section 4.5 (sorry no figure number - something we should add in the=20
> -bis).  Per the spec, there are never any entries that are identical -

> if there, it's a bug. If you think there is text in RFC 4244 that=20
> contradicts this, please identify it and we can clarify it in the
-bis.
> [/MB]

I'll grant I overlooked that example.  The implication is that the tree
presented in History-Info can be *incomplete*, there can be a 1.2 branch
without a 1.1 branch, not because the user used weird numbering, but
because there is a 1.1. branch that is not being described in this
History-Info -- something that is not so common in representations of
trees, and doesn't seem to be explicitly pointed out in the text.

[MB2] Per the spec the only reason to drop entries is due to privacy and
yes, you will know there's been a branch, you just may not know what the
request URI was for that branch nor what the leaves are - kindof like an
incomplete family tree.    And, section 5 (item 3) does highlight the
importance for applications to deal with these situations. So, I'm still
confused as to what text in the current specification is lacking. [/MB2]

In regard to 4.3.3.1.3, the text is not very clear if you don't already
know what it means:

        Retargeting within a Proxy - subsequent instance: For each
        subsequent retargeting of a request by the same proxy, another
        branch is added.  With the index for each new branch calculated
        by incrementing the last/lowest digit at the current level, the
        index in the next request forwarded by this same proxy,
        following the example above, would be 1.1.2.

What is not stated is that the first forked request gets the xxx.1 value
added to its History-Info, the second forked request gets the xxx.2
value added to its History-Info (but not the xxx.1 value), etc.  When I
read this, I assumed that "instance" meant History-Info value, and that
each forked request was expected to have the same set of History-Info
values, namely the whole set generated by the forking, of which the
first "instance" would be numbers xxx.1, etc.

[MB2] I can't see a source of confusion in the RFC - it's talking about
subsequent retargets - i.e., another instance of retargeting and it is
consistent with examples. How would you state it more accurately?
Sorry, I can't grok your point above, so I can't quite see a way to
qualify the text to resolve your concern. [/MB]


> (Assume that sip:1@example.com has been retargeted to=20
> sip:1A@example.com and sip:1B@example.com.)
>=20
> 1.each derived request carries only one of the new History-Info values
>=20
> 	INVITE sip:1A@example.com
> 	History-Info: <sip:1@example.com>;index=3D1,
>                       <sip:1A@example.com>;index=3D1.1
>=20
> 	INVITE sip:1B@example.com
> 	History-Info: <sip:1@example.com>;index=3D1,
>                       <sip:1B@example.com>;index=3D1.1

So the correct answer is case 1, but modified in that the index for
sip:1B@example.com is 1.2; the numbering was globally consistent across
all the History-Info values, but that each value was a subset of the
global forking tree.

[MB2] I'm not clear on why you see case 1 as correct, when the index is
wrong. Yes, in the case of parallel forking that's all the info the
proxy has - i.e., HI can't be more deterministic or provide more
information than what you would get from looking at a message trace.
But, that wouldn't be the case for sequential. [/MB2]

> [MB] I don't see a problem here, as you can indeed determine that the=20
> request was retargetted to User2 as a result of the 3xx - Proxy 1=20
> absolutely knows that and that's the reason why the "reason" is added=20
> to the previous HI entry - i.e., the rule being that the reason is=20
> associated with the "Why the request was retargeted?" from the=20
> previous HI entry (Bob@P2.example.com) to the new HI entry
User@UA2.example.com).
> Thus, you can derive the sequence from the above -i.e., it's ALL there

> in that last INVITE.  It does require that you look for the reason,=20
> but you really don't need the redundant data. An old, old version of=20
> this document actually did have redundant data in that the "old" and
"new"
> were captured in each entry. The current approach is that if the "old"
> isn't captured, then the entity retargeting captures it along with the

> "new" to ensure that you have the entire chain. If you can point to=20
> the specific text that seems to introduce the gap you suggest, that=20
> would be helpful and we can make sure it's clear in the -bis.
> [/MB]

I was thinking of the case where there are two or more forks preceding
the current fork that have returned 3xx.  It seems that you'd get H-I
like:

History-Info:
    <sip:UserA@example.com>;index=3D1,
    <sip:UserA1@example.com?Reason=3DSIP;cause=3D302>;index=3D1.1,
    <sip:UserB@example.com?Reason=3DSIP;cause=3D302>;index=3D1.2,
    <sip:UserC@example.com>;index=3D1.3

>From that, I don't see how to determine whether UserC was generated=20
>from
UserA1 or UserB, or for that matter, if it was derived from UserA.

Perhaps there is an unstated rule that History-Info should contain only
that subset of the global forking tree that describes requests that are
logically ancestral to the request that contains it?

[MB2] I'm not sure of your use of the term "subset" here. If you're
suggesting that only the branches (and not the leaves for a given
branch) are required, that's not really the case UNLESS the privacy
header is being used for some of entries. You can't determine from the
HI entries whether UserC was in the Contact header for the response to
the request forwarded to UserA or UserB OR whether UserC was a 3rd
parallel forked request. The only way to know that for certain would be
to look at the message trace.  It's up to a proxy as to how it might
arbitrate the receipt of the 3xx responses - it is not REQUIRED that
there even be a Contact header in the 3xx. History-Info captures "Why"
the request to UserA and UserB was not successful and NOT why the new
request URI is chosen.    [/MB2]

(Oh, yeah, and the example in RFC 4244 from which I derived the above
example is:

      F12 486 Busy Here Proxy1 -> UA1

      SIP/2.0  486 Busy Here
      Via: SIP/2.0/UDP example.net:5060
      From: Alice <sip:User1@example.net>
      To: Bob <sip:UserA@example.com>
      Call-Id: 12345600@example.net
      History-Info:  <sip:UserA@example.com>; index=3D1,
       <sip:UserA@ims.example.com?Reason=3DSIP;\
       cause=3D302; text=3D"Moved Temporarily">;index=3D1.1,
       <sip:UserB@example.com?Reason=3DSIP;cause=3D480;\
       text=3D"Temporarily Unavailable" >;index=3D1.2,
       <sip:UserC@example.com>;index=3D1.3
      CSeq: 1 INVITE
      Content-Length: 0

Note that it presents a case where one request-URI appears in more than
one History-Info value, which was a question I raised above.)

[MB2] Which request URI in the above example is contained in more than
one HI entry? There are 4 HI entries with 4 different request URIs:
UserA@example.com, UserA@ims.example.com, UserB@example.com and
UserC@example.com   However, it really doesn't matter as there's nothing
that states that the same HI value can't be in multiple entries - it's
the indices that matter - per section  4.3.3.1, this would be just the
case if you are doing loose routing.  [/MB2]

Dale



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Subject: Re: [Sip] draft-ietf-sip-media-security-requirements-09
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> The second last paragraph of section refers to a
> "active-signaling-active-attack-detect" category. Should that not be
> "active-signalling-active-media-detect" ?

Yes, thanks for catching that.  We'll fix it in AUTH48.

-d


From theo@crazygreek.co.uk  Thu Feb 12 18:32:40 2009
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On Fri, Jan 9, 2009 at 9:30 PM, Adam Roach <adam@nostrum.com> wrote:
> I've submitted a new version of draft-roach-sip-http-subscribe, which should
> address most or all of the comments made to date.

Hi Adam,

I really like this version, and will try and find time to implement it
internally over the next few weeks.

in the mean time, a few comments:

2.2, para 3:

   [...] If a server
   wishes to reject such requests, the "403 Forbidden" response code is
   appropriate.

It would be good to provide a way of returning to the client which
ones it was unhappy with in the response, to save it having to go off
and perform a HEAD on each original resource to find out which one has
changed.

2.2, para 5:

  Finally, for each such group, it initiates a subscription to the
group's monitor-list URI;

do you mean "monitor-group" instead of "monitor-list"?

 ~ Theo

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I expect all those who have been holding off revisions due to the RFC 5378 =
issues to start submitting on Monday.

You have of course all got the text ready, and it is just a matter of hitti=
ng the submit button.

regards

Keith

________________________________
From: wgchairs-bounces@ietf.org [mailto:wgchairs-bounces@ietf.org] On Behal=
f Of Ed Juskevicius
Sent: Friday, February 13, 2009 12:53 AM
To: ietf@ietf.org; ietf-announce@ietf.org; wgchairs@ietf.org; iab@iab.org; =
iesg@ietf.org; rfc-editor@rfc-editor.org
Cc: Trustees; Ray Pelletier; Contreras, Jorge
Subject: Re: Last Call for Comments: Proposed work-around to the Pre-5378 P=
roblem

I am pleased to announce that the IETF Trustees have just finished work on =
a revision to our "Legal Provisions Relating to IETF Documents" policy.  Th=
e revision includes optional new legend text in Section 6.c.iii of the poli=
cy to serve as a work-around for people experiencing the "pre-5378 problem.=
"

Please note the newly approved policy is dated 2009-02-12.  Please also not=
e that the Effective Date of this revised policy has been set to 2009-02-15=
.  The old policy (effective from November 10, 2008) remains in force until=
 00h00 UTC on February 15th.

The approved text is available now on IETF Trust website at
 http://trustee.ietf.org/policyandprocedures.html

Please look for the document dated 2009-02-12, just below the heading label=
ed "DRAFT Policy and Procedures Being Developed."

On or shortly after February 15th, the Trust website will be updated so as =
to archive all of the recent draft versions of the new policy, and to make =
it easier to navigagte to the newly approved policy.

Please review the new policy so you are aware of the latest copyright state=
ments and other boilerplate legend text that will be required on all contri=
butions to the IETF starting on February 15, 2009.

Regards, and Thanks in advance,


Ed Juskevicius, on behalf of the IETF Trustees
edj.etc@gmail.com<mailto:edj.etc@gmail.com>

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Diso-8859-1"=
>
<META content=3D"MSHTML 6.00.2900.3492" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009>I expect all those who have been holding off rev=
isions=20
due to the RFC 5378 issues to start submitting on Monday.</SPAN></FONT></DI=
V>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009>You have of course all got the text ready, and i=
t is=20
just a matter of hitting the submit button.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009>regards</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D664331405-16022009>Keith</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> wgchairs-bounces@ietf.org=20
[mailto:wgchairs-bounces@ietf.org] <B>On Behalf Of </B>Ed=20
Juskevicius<BR><B>Sent:</B> Friday, February 13, 2009 12:53 AM<BR><B>To:</B=
>=20
ietf@ietf.org; ietf-announce@ietf.org; wgchairs@ietf.org; iab@iab.org;=20
iesg@ietf.org; rfc-editor@rfc-editor.org<BR><B>Cc:</B> Trustees; Ray Pellet=
ier;=20
Contreras, Jorge<BR><B>Subject:</B> Re: Last Call for Comments: Proposed=20
work-around to the Pre-5378 Problem<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>I am pleased to announce that the IETF Trustees have&nbsp;just finishe=
d=20
work on a revision to&nbsp;our&nbsp;"Legal Provisions Relating to IETF=20
Documents" policy.&nbsp;&nbsp;The revision includes optional new&nbsp;legen=
d=20
text in Section 6.c.iii of the policy&nbsp;to&nbsp;serve as a work-around f=
or=20
people experiencing the "pre-5378 problem."</DIV>
<DIV>&nbsp;</DIV>
<DIV>Please note the newly approved&nbsp;policy is dated 2009-02-12.&nbsp;=
=20
Please also note that&nbsp;the Effective Date of this revised policy&nbsp;h=
as=20
been set to&nbsp;2009-02-15.&nbsp; The old policy=20
(effective&nbsp;from&nbsp;November 10, 2008)&nbsp;remains in&nbsp;force unt=
il=20
00h00 UTC on February 15th.</DIV>
<DIV>&nbsp;</DIV>
<DIV>The approved text is available&nbsp;now on&nbsp;IETF Trust website at<=
/DIV>
<DIV>&nbsp;<A=20
href=3D"http://trustee.ietf.org/policyandprocedures.html">http://trustee.ie=
tf.org/policyandprocedures.html</A><BR><BR>Please=20
look for the&nbsp;document&nbsp;dated 2009-02-12, just below the&nbsp;headi=
ng=20
labeled "DRAFT Policy and Procedures Being Developed."</DIV>
<DIV>&nbsp;</DIV>
<DIV>On or shortly after February 15th, the Trust website will be updated s=
o as=20
to archive all of the recent draft versions of the new policy, and to&nbsp;=
make=20
it easier to navigagte to the&nbsp;newly approved policy.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Please review the new policy so you are aware of the latest copyright=
=20
statements and other boilerplate legend text that&nbsp;will be required on=
=20
all&nbsp;contributions to the IETF starting on February 15, 2009.&nbsp;</DI=
V>
<DIV>&nbsp;</DIV>
<DIV>Regards, and Thanks in advance,</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>Ed Juskevicius, on behalf of the IETF Trustees</DIV>
<DIV><FONT color=3D#888888><FONT color=3D#888888><A href=3D"mailto:edj.etc@=
gmail.com"=20
target=3D_blank>edj.etc@gmail.com</A></FONT><BR></FONT></DIV></BODY></HTML>

--_000_28B7C3AA2A7ABA4A841F11217ABE78D674A08C20FRMRSSXCHMBSB3d_--

From dean.willis@softarmor.com  Mon Feb 16 11:07:42 2009
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Subject: [Sip] WGLC on draft-ietf-sip-info-events
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I'd like to request the SIP Working Group perform final review on the  
INFO Events draft:


http://tools.ietf.org/id/draft-ietf-sip-info-events-03.txt



Please review this draft and bring any issues to the mailing list,  
copying the author.  Please conclude this review process by March 4,  
2009.

--
Dean Willis
cp-Chair, SIP WG

From drage@alcatel-lucent.com  Mon Feb 16 16:00:54 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: SIP List <SIP@ietf.org>
Date: Tue, 17 Feb 2009 01:00:55 +0100
Thread-Topic: WGLC for draft-ietf-sip-ua-privacy
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Cc: "draft-ietf-sip-ua-privacy@tools.ietf.org" <draft-ietf-sip-ua-privacy@tools.ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: [Sip] WGLC for draft-ietf-sip-ua-privacy
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(As WG cochair)

This mail is to initiate working group last call on draft-ietf-sip-ua-priva=
cy which may be found here:

http://tools.ietf.org/internet-drafts/draft-ietf-sip-ua-privacy-05.txt

Please provide comments to both the authors and to the SIP WG list by Monda=
y 2nd March 2009.

The intended status of the document is informational.

regards

Keith=

From brett@broadsoft.com  Mon Feb 16 16:18:26 2009
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From: Brett Tate <brett@broadsoft.com>
To: SIP <sip@ietf.org>, "draft-ietf-sip-info-events@tools.ietf.org" <draft-ietf-sip-info-events@tools.ietf.org>
Date: Mon, 16 Feb 2009 16:18:28 -0800
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Howdy,

The following are a few comments and questions concerning draft-ietf-sip-in=
fo-events-03.

Thanks,
Brett

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

Section 3.1 paragraph 1:  Concerning "MUST advertise" and "provisional 1xx"=
, please clarify that the "MUST advertise" is not applicable to the 100 res=
ponse.  Does the "MUST advertise" and "provisional 1xx" apply to all (first=
 and subsequent within dialog) INVITE non 101-199 (intentionally indicated =
199)?  Concerning ACK for initial INVITE's 2xx, does the "MUST advertise" a=
lways apply?

Section 3.1 paragraph 2 sentences 1 and 2: Sentence 1 indicates "session pa=
rameters" and sentence 2 indicates "dialog parameters" concerning Info Pack=
age negotiation, I'm not sure when the negotiation/renegotiation occurs.  D=
oes it follow SDP offer/answer rules or something else; if something else, =
what?  I have the same question concerning "dialog refresh" text discussed =
later within the same section since incorrect interpretations might trigger=
 deactivation.

Section 4.3: When sending a 489, it indicates "MUST terminate the INVITE di=
alog".  The "MUST terminate" seems severe even if all the negotiation race =
conditions are adequately addressed.  The mentioned "protocol failure" reas=
on seems contrary to the "strictly send and lenient receive" mantra.

Section 5.1 Table 1: The usage of "g" seems potentially outdated or incorre=
ctly reference since discussed within RFC 2543 instead of RFC 3261.  The sa=
me applies to usage Via row concerning "(2)".

Section 5.2.2: The "general-header" reference is outdated or incorrectly re=
ference since defined within RFC 2543 instead of RFC 3261.  RFC 3261 uses "=
Message-Header".

Section 7.7 paragraph 2: The following potentially could be reworded since =
it is okay to return failure responses (401, 503, 500, ...) even when overl=
apping occurs: "in the event a UAC issues overlapping INFO requests for an =
Info Package, the UAS MUST NOT return an error response".

Section 8: Why specifically defining and using "nil"?  I'm mainly just curi=
ous of the benefit over not having defined "nil" for similar use within the=
 Supported header.


From michael@voip.co.uk  Tue Feb 17 00:51:05 2009
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Just a minor point:  Is it worth adding (either in section 4.1 or 6)
that a temp-gruu might not be as anonymous as you might hope?  An
observer using RFC 3680 (reg-event) with gruu extensions would be able
to correlate temp-gruus with AoRs and contacts, should they be so
authorised.

There is some text in RFC 3680 warning of the risks of reg-event, but
that is probably of more direct interest to registrar authors.  A
reminder of the risk in this document might highlight it for UA authors,
so that they can consider the wider implications.

Best regards,

Michael

From christer.holmberg@ericsson.com  Tue Feb 17 05:49:21 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

I've submitted a new version of draft-holmberg-sip-keep. The draft is
more or less identical to the previous one, with an open issue removed.

I actually thought that we reached concensus of some of the use-cases
(not the actual mechanism) presented at the MSP meeting, and that we
were going to start working on this. But, obviously I was wrong, because
no milestone has been created for this...

So, I will now need some guidance from the chairs and/or ADs on what to
do next...

You can also find the draft at:

http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-keep-03.txt

Regards,

Christer

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

<P><FONT SIZE=3D2 FACE=3D"Arial">I've submitted a new version of =
draft-holmberg-sip-keep. The draft is more or less identical to the =
previous one, with an open issue removed.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I actually thought that we reached =
concensus of some of the use-cases (not the actual mechanism) presented =
at the MSP meeting, and that we were going to start working on this. =
But, obviously I was wrong, because no milestone has been created for =
this...</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">So, I will now need some guidance from =
the chairs and/or ADs on what to do next...</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">You can also find the draft at:</FONT>
</P>

<P><A =
HREF=3D"http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-keep-03=
.txt"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-=
keep-03.txt</FONT></U></A>
</P>

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

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

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From drage@alcatel-lucent.com  Tue Feb 17 17:53:13 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Michael Procter <michael@voip.co.uk>, "draft-ietf-sip-ua-privacy@tools.ietf.org" <draft-ietf-sip-ua-privacy@tools.ietf.org>, "sip@ietf.org" <sip@ietf.org>
Date: Wed, 18 Feb 2009 02:53:17 +0100
Thread-Topic: [Sip] Comment on sip-ua-privacy-05.txt
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Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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There is a similar issue with the instance-id as used by outbound and GRUU =
as well. In a mobile, this uses the IMEI which is not necessarily meant to =
be revealed.

3GPP obfuscate the IMEI on generation of the GRUU.

regards

Keith=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Michael Procter
> Sent: Tuesday, February 17, 2009 8:51 AM
> To: draft-ietf-sip-ua-privacy@tools.ietf.org; sip@ietf.org
> Subject: [Sip] Comment on sip-ua-privacy-05.txt
>=20
> Just a minor point:  Is it worth adding (either in section=20
> 4.1 or 6) that a temp-gruu might not be as anonymous as you=20
> might hope?  An observer using RFC 3680 (reg-event) with gruu=20
> extensions would be able to correlate temp-gruus with AoRs=20
> and contacts, should they be so authorised.
>=20
> There is some text in RFC 3680 warning of the risks of=20
> reg-event, but that is probably of more direct interest to=20
> registrar authors.  A reminder of the risk in this document=20
> might highlight it for UA authors, so that they can consider=20
> the wider implications.
>=20
> Best regards,
>=20
> Michael
> _______________________________________________
> Sip mailing list  https://www.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=20
> Use sipping@ietf.org for new developments on the application of sip
> =

From shida@ntt-at.com  Wed Feb 18 23:34:38 2009
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Cc: "sip@ietf.org" <sip@ietf.org>, draft-ietf-sip-ua-privacy@tools.ietf.org
Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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Thanks for pointing this out..

  I went and looked at gruu-reg-event draft and saw what kind
of security consideration text is there, the text described a  
authorization
policy limiting only allowing subscription to those who actually  
registered
and requested the temp-gruu.

  I do agree that implication of reg-event and some consideration for
the implementors will be beneficial and will add some text in the  
security
consideration.

  Many Thanks
   Shida

On 17-Feb-09, at 5:51 PM, Michael Procter wrote:

> Just a minor point:  Is it worth adding (either in section 4.1 or 6)
> that a temp-gruu might not be as anonymous as you might hope?  An
> observer using RFC 3680 (reg-event) with gruu extensions would be able
> to correlate temp-gruus with AoRs and contacts, should they be so
> authorised.
>
> There is some text in RFC 3680 warning of the risks of reg-event, but
> that is probably of more direct interest to registrar authors.  A
> reminder of the risk in this document might highlight it for UA  
> authors,
> so that they can consider the wider implications.
>
> Best regards,
>
> Michael
> _______________________________________________
> Sip mailing list  https://www.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 shida@ntt-at.com  Wed Feb 18 23:52:41 2009
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Cc: "sip@ietf.org" <sip@ietf.org>, "draft-ietf-sip-ua-privacy@tools.ietf.org" <draft-ietf-sip-ua-privacy@tools.ietf.org>
Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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Keith;

  I just looked at outbound and gruu again, and AFAICT temp-gruu doesn't
carry the instance-id unlike the pub-gruu which does.

  With outbound the instance-id is only present in the REGISTER
request when you are establishing the flow, and if temp-gruu is used,
the only time instance-id will be visible when GRUU is used is also
in the REGISTER request.

  I may be blind but I can't seem to see the problem.

  Regards
   Shida

On 18-Feb-09, at 10:53 AM, DRAGE, Keith (Keith) wrote:

> There is a similar issue with the instance-id as used by outbound  
> and GRUU as well. In a mobile, this uses the IMEI which is not  
> necessarily meant to be revealed.
>
> 3GPP obfuscate the IMEI on generation of the GRUU.
>
> regards
>
> Keith
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>> Behalf Of Michael Procter
>> Sent: Tuesday, February 17, 2009 8:51 AM
>> To: draft-ietf-sip-ua-privacy@tools.ietf.org; sip@ietf.org
>> Subject: [Sip] Comment on sip-ua-privacy-05.txt
>>
>> Just a minor point:  Is it worth adding (either in section
>> 4.1 or 6) that a temp-gruu might not be as anonymous as you
>> might hope?  An observer using RFC 3680 (reg-event) with gruu
>> extensions would be able to correlate temp-gruus with AoRs
>> and contacts, should they be so authorised.
>>
>> There is some text in RFC 3680 warning of the risks of
>> reg-event, but that is probably of more direct interest to
>> registrar authors.  A reminder of the risk in this document
>> might highlight it for UA authors, so that they can consider
>> the wider implications.
>>
>> Best regards,
>>
>> Michael
>> _______________________________________________
>> Sip mailing list  https://www.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://www.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 web-usrn@ISI.EDU  Thu Feb 19 03:02:14 2009
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To: srd@cisco.com, jdrosen@cisco.com, jon.peterson@neustar.biz, fluffy@cisco.com, dean.willis@softarmor.com, drage@alcatel-lucent.com
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Subject: [Sip] [Technical Errata Reported] RFC4028 (1687)
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The following errata report has been submitted for RFC4028,
"Session Timers in the Session Initiation Protocol (SIP)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=4028&eid=1687

--------------------------------------
Type: Technical
Reported by: Radha krishna Saragadam <rsaragad@cisco.com>

Section: 9

Original Text
-------------
The UAS MUST NOT increase the value of the Session-Expires header field.


Corrected Text
--------------
same as session 8.1
   If the request doesn't indicate support for the session timer but
   contains a session interval that is too small, the UAS cannot
   usefully reject the request, as this would result in a call failure.
   Rather, the UAS SHOULD insert a Min-SE header field containing its
   minimum interval.  If a Min-SE header field is already present, the
   UAS SHOULD increase (but MUST NOT decrease) the value to its
   minimum interval.  The UAS MUST then increase the Session-Expires
   header field value to be equal to the value in the Min-SE header
   field

Notes
-----
----- Forwarded Message ----
From: Radha krishna <krishna_srk2003@yahoo.com>
To: Brett Tate <brett@broadsoft.com>; "sip-implementors@lists.cs.columbia.edu" <sip-implementors@lists.cs.columbia.edu>
Sent: Thursday, February 19, 2009 10:56:31 AM
Subject: Re: [Sip-implementors] Sending 422

Thanks Brett,

      So I think same should be added for UAS

<Snip from RFC section 8.1>

  If the request doesn't indicate support for the session timer but
 
contains a session interval that is too small, the proxy cannot
  usefully
reject the request, as this would result in a call failure.
  Rather, the
proxy SHOULD insert a Min-SE header field containing its
  minimum
interval.  If a Min-SE header field is already present, the
  proxy SHOULD
increase (but MUST NOT decrease) the value to its
  minimum interval.  The
proxy MUST then increase the Session-Expires
  header field value to be equal to the value in the
Min-SE header
  field, as described above.
</Snip from RFC>


Regards
S.Radha krishna




________________________________
From: Brett Tate <brett@broadsoft.com>
To: Radha krishna <krishna_srk2003@yahoo.com>; "sip-implementors@lists.cs.columbia.edu" <sip-implementors@lists.cs.columbia.edu>
Sent: Wednesday, February 18, 2009 6:42:31 PM
Subject: RE: [Sip-implementors] Sending 422

It looks like Section 9 may have forgotten to indicate the behavior when UAC timer support not indicated.  Section 8.1 allows a proxy to increase the Session-Expires; I see no reason why the same cannot be done by the UAS.

> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-
> implementors-bounces@lists.cs.columbia.edu] On Behalf Of Radha krishna
> Sent: Tuesday, February 17, 2009 9:48 PM
> To: sip-implementors@lists.cs.columbia.edu
> Subject: [Sip-implementors] Sending 422
>
> Hi
>
>          Consider the following topology
>              UA1 ----- Call-stateful-proxy ------ UA2
>
>          UA1 does not support session timer, Make a call to UA2. Call-
> stateful-proxy adds Session-Expires:100 header and forwards to UA2. UA2
> minimum session expires is 900. But in this case INVITE will not contain
> "support: timer". According section 9, UAS can reject with 422 only if
> there is a timer tag in supported header
>
> <Snip from RFC>
>
>    If an incoming request contains a Supported header field with a value
>    'timer' and a Session Expires header field, the UAS MAY reject the
>    INVITE request with a 422 (Session Interval Too Small) response if
>    the session interval in the Session-Expires header field is smaller
>    than the minimum interval defined by the UAS' local policy.  When
>    sending the 422 response, the UAS MUST include a Min-SE header field
>    with the value of its minimum interval.  This minimum interval MUST
>    NOT be lower than 90 seconds.
> </Snip from RFC>
>
>        Also UAS cannot increase the session expires duration
> <Snip from RFC>
> The UAS MUST
>    NOT increase the value of the Session-Expires header field.
> </Snip from RFC>
>
> What should be the behavior of UAS here?
>    1) accept the call with 100 seconds?
>    2) Increase the duration to 900 seconds while sending 200 Ok?
> Note: Session timer should not be turned-off
>
> Regards
> S.Radha krishna
>
>
>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors



     
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--------------------------------------
RFC4028 (draft-ietf-sip-session-timer-15)
--------------------------------------
Title               : Session Timers in the Session Initiation Protocol (SIP)
Publication Date    : April 2005
Author(s)           : S. Donovan, J. Rosenberg
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From christer.holmberg@ericsson.com  Thu Feb 19 03:11:53 2009
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Thread-Topic: New version (-05) of the 199 draft - ready for WGLC?
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

I've submitted a new version of draft-ietf-sip-199. I have done changes
and updates based on feedback and discussions (please let me know if you
think I've forgotten something), and I've also added text to the
security chapter.

I propose that we start the WGLC process for the draft.

The draft can also be found at:

http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt

Regards,

Christer

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

<P><FONT SIZE=3D2 FACE=3D"Arial">I've submitted a new version of =
draft-ietf-sip-199. I have done changes and updates based on feedback =
and discussions (please let me know if you think I've forgotten =
something), and I've also added text to the security chapter.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I propose that we start the WGLC =
process for the draft.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The draft can also be found at:</FONT>
</P>

<P><A =
HREF=3D"http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt"=
><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-=
05.txt</FONT></U></A>
</P>

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

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


	Title           : Response Code for Indication of Terminated Dialog
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sip-199-05.txt
	Pages           : 14
	Date            : 2009-02-19

This specification defines a new SIP response code, 199 Early Dialog
Terminated, which a SIP forking proxy and a UAS can use to indicate
upstream towards the UAC that an early dialog has been terminated,
before a final response is sent towards the UAC.

A URL for this Internet-Draft is:
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From: "Cowby" <cnmjbm@gmail.com>
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Subject: [Sip] questions about usage of tel URI with a SIP UA
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Hi,

I have few question about "tel" uri defined in RFC 3966.

Q1: is it possible that a SIP UA can receive incoming requests with tel uri
as the request-uri?

Q2: if the answer to Q1 is yes, how does a SIP UA normally handle tel uri to
match the UA itself?

For example, if a UA has a mailto:"sip:cowby@cows.com' as its IMS public
identity and +1-408-555-6666 as it global phone number,

Q1: is it possible that the UA's P-CSCF can route an INVITE
tel:+1-408-555-6666 request to the UA?
Q2: how can the UA match itself when it is configured as cowby@cows.com but
know nothing of its phone number?

Thanks.

Cowby


From theo@crazygreek.co.uk  Thu Feb 19 20:20:00 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Hi,

I've just submitted an ID that discusses the inherent UDP "problem",
along with a potential workaround to stop it being used maliciously.

I would really welcome any feedback:

  http://www.ietf.org/internet-drafts/draft-zourzouvillys-sip-via-cookie-00.txt

 ~ Theo

From shamik.saha@wipro.com  Thu Feb 19 22:13:22 2009
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Hi Theo,

I was just thinking of a scenario,suppose the attacker creates a spoofed
invite,and inserts his ip in the Via header field and an empty cookie
parameter in the via header field.The Proxie which he wants to use as an
ampilfier in a DoS attack,on receiving such a request will respond with
a 4xx response,and the value of the cookie parameter to be used.On
receiving the cookie,the attacker,resends the invite with the cookie
value and deletes its entry from the Via header field.Thus now the
responses for this invite will be sent back to the UAC which is to be
the victim of the DoS attack only.The UAC which is attacked will receive
a minimum of 22 responses only.

What do you think about this.


Thanks and regards,
=20
Shamik Saha
Project Engineer
Voice Protocols
Cell :  +91-9886704155

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Theo Zourzouvillys
Sent: Friday, February 20, 2009 9:50 AM
To: SIP WG
Subject: [Sip] draft-zourzouvillys-sip-via-cookie-00

Hi,

I've just submitted an ID that discusses the inherent UDP "problem",
along with a potential workaround to stop it being used maliciously.

I would really welcome any feedback:

=20
http://www.ietf.org/internet-drafts/draft-zourzouvillys-sip-via-cookie-0
0.txt

 ~ Theo
_______________________________________________
Sip mailing list  https://www.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: Fri, 20 Feb 2009 07:43:15 +0000
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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On Fri, Feb 20, 2009 at 6:03 AM,  <shamik.saha@wipro.com> wrote:
> I was just thinking of a scenario,suppose the attacker creates a spoofed
> invite,and inserts his ip in the Via header field and an empty cookie
> parameter in the via header field.The Proxie which he wants to use as an
> ampilfier in a DoS attack,on receiving such a request will respond with
> a 4xx response,and the value of the cookie parameter to be used.On
> receiving the cookie,the attacker,resends the invite with the cookie
> value and deletes its entry from the Via header field.Thus now the
> responses for this invite will be sent back to the UAC which is to be
> the victim of the DoS attack only.The UAC which is attacked will receive
> a minimum of 22 responses only.

SIP responses will always be sent back to the source of the request
(and if rport is included also to the source port), and not the value
of sent-by in the first instance.  The cookie the server generates
would always use the source ip and port in the cookie it creates, so
sending the second request with a different source IP would mean the
server can detect the cookie is invalid.

However, an attacker could indeed make sent-by the address of the
victim, perform the cookie dance (legitimately), and then play dead by
sending ICMP messages for any responses received. However, following
rfc3261 sect 18.2.2 and rfc3263 to the last dot would result in the
response being transmitted every time to the source (attacker) first
before then being transmitted to the victim (after an ICMP port/host
unreachable response), therefore making it a highly unattractive
proposition for the attacker.

 ~ Theo

From theo@crazygreek.co.uk  Thu Feb 19 23:45:29 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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On Fri, Feb 20, 2009 at 4:20 AM, Theo Zourzouvillys
<theo@crazygreek.co.uk> wrote:

> I've just submitted an ID that discusses the inherent UDP "problem",
> along with a potential workaround to stop it being used maliciously.

note that it's *only* trying to solve the drive-by-shooting problem
that is inherent in UDP where an attacker can just "fire and forget"
by spoofing the source;  it does _not_ try to solve the problem of
verifying Contact or Route header values when establishing a dialog.
I believe this is a separate issue (as it affects all transports, not
just UDP), and am only trying to bring UDP onto the same footing as
connection oriented transports.

Having just read through the draft it again, it would undoubtedly make
more sense to use a non 2xx response as an example instead, and remove
the mention of 100rel to make it clear i'm not trying to solve that :)

 ~ Theo

From lists@ohlmeier.org  Fri Feb 20 00:58:41 2009
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Subject: Re: [Sip] questions about usage of tel URI with a SIP UA
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Hello,

Am 20.02.2009 2:05 Uhr, schrieb Cowby:
> Q1: is it possible that a SIP UA can receive incoming requests with tel uri
> as the request-uri?

In IMS yes.

> Q2: if the answer to Q1 is yes, how does a SIP UA normally handle tel uri to
> match the UA itself?

Take a look at the P-Associated-URI header.

> For example, if a UA has a mailto:"sip:cowby@cows.com' as its IMS public
> identity and +1-408-555-6666 as it global phone number,
>
> Q1: is it possible that the UA's P-CSCF can route an INVITE
> tel:+1-408-555-6666 request to the UA?

Yes, see above.

> Q2: how can the UA match itself when it is configured as cowby@cows.com but
> know nothing of its phone number?

See above as well.

BTW such questions should be directed to the sip-implementors mailing 
list instead (although I'm not sure if they discuss IMS as well).

Best regards
   Nils Ohlmeier

From lists@ohlmeier.org  Fri Feb 20 01:46:22 2009
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Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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Hi Theo,

Am 20.02.2009 5:20 Uhr, schrieb Theo Zourzouvillys:
> I've just submitted an ID that discusses the inherent UDP "problem",
> along with a potential workaround to stop it being used maliciously.
>
> I would really welcome any feedback:
>
>    http://www.ietf.org/internet-drafts/draft-zourzouvillys-sip-via-cookie-00.txt

in general I like your proposal, but I have a few questions/thoughts:

I think the whole "attack scenario" which is described in the draft 
works only if the attacker uses the proxy/registrar of the victim as a 
relay.
Reason 1): because the attacker hopefully does not know which UDP port 
of the victim is currently "open" to accept incoming UDP packets. And if 
their is a NAT or firewall in between the UA it will hopefully only 
accept UDP packets from the proxy/registrar. If the UA of the victim 
runs on a public IP without any "protection" it might accept SIP 
requests from anywhere, although I would call this a very bad UA 
implementation.
Reason 2): when the UA of the victim registeres itself it provides a 
Contact URI. I general I believe UA should only accept incoming requests 
from their registrar IP and port which contains the Contact URI which 
they provided at the registration. This prevents very easily that 
someone sends drive by requests just with the AOR of the victim as R-URI 
to the UA of the victim. If a registrar forwards requests to registered 
UAs with the AOR of the user in the R-URI the registrar is broken in my 
opinion.

Now the question is of the proxy/registrar accepts requests from its 
users from anywhere. I guess that is what you mean with no 
authentication, right?

One bigger concern for the whole approach is: if the attacker knows 
about this cookie extension he will for sure not put an empty cookie in 
the Via header of the request. But how does the proxy of the victim 
handle requests without the cookie extension?
To be backward compatible it would need to accept requests without the 
extension, but then the door is open for the attacker again.
If the proxy of the victim does not care about backward compatibility 
(maybe because attack previntion is more important), wouldn't the need a 
way to tell the sender of the request that the cookie extension is 
required to successfully relay requests? So wouldn't we need a new value 
for the Required header, or would a new 4XX response already be enough.

As I said I like the general idea. But I believe that the attack 
scenario which you describe in your draft should be a little bit more 
narrow then described.

Best regards
   Nils Ohlmeier

From hsinnrei@adobe.com  Fri Feb 20 06:44:49 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Nils Ohlmeier <lists@ohlmeier.org>, Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Fri, 20 Feb 2009 06:44:40 -0800
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--_000_C5C41DF8B712hsinnreiadobecom_
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Nils,

>If the UA of the victim runs on a public IP without any "protection"
>it might accept SIP requests from anywhere, although
>I would call this a very bad UA implementation.

This is correct.

Why cannot the UA have adequate protection at the application layer, such a=
s authenticating the caller?

Making a massive DOS attack on one single UAs does not seem a "productive" =
technique for spammers.
And if one individual is a valuable target, such individuals have other com=
munication means besides a sole SIP UA.

Henry


On 2/20/09 3:46 AM, "Nils Ohlmeier" <lists@ohlmeier.org> wrote:

If the UA of the victim
runs on a public IP without any "protection" it might accept SIP
requests from anywhere, although I would call this a very bad UA
implementation.

--_000_C5C41DF8B712hsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [Sip] draft-zourzouvillys-sip-via-cookie-00</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
14pt'>Nils,<BR>
<BR>
&gt;If the UA of the victim runs on a public IP without any &quot;protectio=
n&quot; <BR>
&gt;it might accept SIP requests from anywhere, although <BR>
&gt;I would call this a very bad UA implementation.<BR>
<BR>
This is correct.<BR>
<BR>
Why cannot the UA have adequate protection at the application layer, such a=
s authenticating the caller?<BR>
<BR>
Making a massive DOS attack on one single UAs does not seem a &#8220;produc=
tive&#8221; technique for spammers.<BR>
And if one individual is a valuable target, such individuals have other com=
munication means besides a sole SIP UA.<BR>
<BR>
Henry <BR>
<BR>
<BR>
On 2/20/09 3:46 AM, &quot;Nils Ohlmeier&quot; &lt;<a href=3D"lists@ohlmeier=
.org">lists@ohlmeier.org</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:14pt'> If the UA of the victim<BR>
runs on a public IP without any &quot;protection&quot; it might accept SIP<=
BR>
requests from anywhere, although I would call this a very bad UA<BR>
implementation.<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


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From theo@crazygreek.co.uk  Fri Feb 20 07:37:06 2009
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On Fri, Feb 20, 2009 at 9:46 AM, Nils Ohlmeier <lists@ohlmeier.org> wrote:

Hi Nils,

Thanks for the feedback - i'll elaborate to make the issues you raised
more clear in the next version of the document.

> I think the whole "attack scenario" which is described in the draft works
> only if the attacker uses the proxy/registrar of the victim as a relay.
> Reason 1): because the attacker hopefully does not know which UDP port of
> the victim is currently "open" to accept incoming UDP packets. And if their
> is a NAT or firewall in between the UA it will hopefully only accept UDP
> packets from the proxy/registrar. If the UA of the victim runs on a public
> IP without any "protection" it might accept SIP requests from anywhere,
> although I would call this a very bad UA implementation.

A victim does not need to have an open UDP port;  An attack can easily
be launched against any IP address, even a non SIP related one.

A device behind NAT on the end of a link additionally has no
protection at an IP level, as the link would easily become saturated
as a reflector starts sending failure response retransmits from a IST
every time timerG fires.

> Reason 2): when the UA of the victim registeres itself it provides a Contact
> URI. I general I believe UA should only accept incoming requests from their
> registrar IP and port which contains the Contact URI which they provided at
> the registration. This prevents very easily that someone sends drive by
> requests just with the AOR of the victim as R-URI to the UA of the victim.
> If a registrar forwards requests to registered UAs with the AOR of the user
> in the R-URI the registrar is broken in my opinion.

as above - a victim does not need to be a SIP UA - it doesn't even
need to talk SIP.  The attacker may well be trying to attack a non SIP
entity, for example a web, DNS, or mail server.

On the flip side, a UA that "only accept"s requests from it's
registrar if of course good practice, but is only a step in stopping
SPIT, not denial of service (which this draft tries to tackle), unless
the "only accept" is limited by routability.  Note even NAT doesn't
stop a denial of service attack being launched against the NAT
gateway, causing it's connectivity to become saturated.

This draft is specifically trying to stopping a SIP server transaction
being used as an amplifying reflector in a (distributed) denial of
service attack.  The victim of the attack does not need to be a SIP
endpoint (it doesn't even need to have any ports open!)

> Now the question is of the proxy/registrar accepts requests from its users
> from anywhere. I guess that is what you mean with no authentication, right?

Yes - any SIP server (note a UAS or proxy are both servers for this)
which allows an unauthenticated, untrusted user to create an invite
server transaction (even for failure delivery) is vulnerable.

> One bigger concern for the whole approach is: if the attacker knows about
> this cookie extension he will for sure not put an empty cookie in the Via
> header of the request. But how does the proxy of the victim handle requests
> without the cookie extension?

Any proxy that does wish to be used as an amplifier should reject
requests without cookies with a stateless 4XX, or send a stateless
401/407 if knows it can authenticate it's user.

Of course this is not going to be realistic from day 1, but this
document is a start in finding solutions to some of the DOS problems
we're currently susceptible to by putting a SIP server "on the net".

Right now, there are a large number of SIP endpoints out there that
will amplify a single UDP packet sent to them, and the initiator of
the attack isn't even traceable!

(i'm starting to sound like a scaremonger, so i'll stop now :))

> If the proxy of the victim does not care about backward compatibility (maybe
> because attack previntion is more important), wouldn't the need a way to
> tell the sender of the request that the cookie extension is required to
> successfully relay requests? So wouldn't we need a new value for the
> Required header, or would a new 4XX response already be enough.

I did briefly consider an option tag but the problem is a SIP UA does
not need to implement anything for it to potentially work - for
example a proxy can implement it on behalf of a group of trusted users
(for example, a corporate proxy or an outbound proxy that has
authenticated the inbound request).

There would additionally be no mechanism for recovery if the server
refuses to process UDP requests without cookie support.  The 4XX would
provide a code that allows a machine to understand that it failed due
to cookies (if it knew about, but didn't implement the extension) -
and could for example try upgrading to TCP, [D]TLS etc.  The reason
phrase provides something descriptive for devices that don't
understand it.  For those that don't, it'll fall back to treating it
like a 400.

 ~ Theo

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On Fri, Feb 20, 2009 at 2:44 PM, Henry Sinnreich <hsinnrei@adobe.com> wrote=
:

Hi Henry,

> Making a massive DOS attack on one single UAs does not seem a =93producti=
ve=94
> technique for spammers.

The problem isn't trying to stop spammers at all, it's trying to stop
SIP servers being used as DOS amplifiers.

> And if one individual is a valuable target, such individuals have other
> communication means besides a sole SIP UA.

The victim does not need to be a SIP UA, it could be any IP address at all.

 ~ Theo

From theo@crazygreek.co.uk  Fri Feb 20 07:42:07 2009
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Date: Fri, 20 Feb 2009 15:42:18 +0000
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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On Fri, Feb 20, 2009 at 3:36 PM, Theo Zourzouvillys
<theo@crazygreek.co.uk> wrote:
> Any proxy that does wish to be used as an amplifier should reject
> requests without cookies with a stateless 4XX, or send a stateless
> 401/407 if knows it can authenticate it's user.

I of course mean does NOT wish to be used, ahem :-)

 ~ Theo

From hsinnrei@adobe.com  Fri Feb 20 07:45:14 2009
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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>
Date: Fri, 20 Feb 2009 07:45:08 -0800
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--_000_C5C42C24B724hsinnreiadobecom_
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>it's trying to stop SIP servers being used as DOS amplifiers.

If you take this viewpoint, then the fewer servers, the less vulnerability?

Henry


On 2/20/09 9:40 AM, "Theo Zourzouvillys" <theo@crazygreek.co.uk> wrote:

On Fri, Feb 20, 2009 at 2:44 PM, Henry Sinnreich <hsinnrei@adobe.com> wrote=
:

Hi Henry,

> Making a massive DOS attack on one single UAs does not seem a "productive=
"
> technique for spammers.

The problem isn't trying to stop spammers at all, it's trying to stop
SIP servers being used as DOS amplifiers.

> And if one individual is a valuable target, such individuals have other
> communication means besides a sole SIP UA.

The victim does not need to be a SIP UA, it could be any IP address at all.

 ~ Theo


--_000_C5C42C24B724hsinnreiadobecom_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [Sip] draft-zourzouvillys-sip-via-cookie-00</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
14pt'>&gt;it's trying to stop SIP servers being used as DOS amplifiers.<BR>
<BR>
If you take this viewpoint, then the fewer servers, the less vulnerability?=
<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 2/20/09 9:40 AM, &quot;Theo Zourzouvillys&quot; &lt;<a href=3D"theo@craz=
ygreek.co.uk">theo@crazygreek.co.uk</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:14pt'>On Fri, Feb 20, 2009 at 2:44 PM, Henry Sinn=
reich &lt;<a href=3D"hsinnrei@adobe.com">hsinnrei@adobe.com</a>&gt; wrote:<=
BR>
<BR>
Hi Henry,<BR>
<BR>
&gt; Making a massive DOS attack on one single UAs does not seem a &#8220;p=
roductive&#8221;<BR>
&gt; technique for spammers.<BR>
<BR>
The problem isn't trying to stop spammers at all, it's trying to stop<BR>
SIP servers being used as DOS amplifiers.<BR>
<BR>
&gt; And if one individual is a valuable target, such individuals have othe=
r<BR>
&gt; communication means besides a sole SIP UA.<BR>
<BR>
The victim does not need to be a SIP UA, it could be any IP address at all.=
<BR>
<BR>
&nbsp;~ Theo<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C5C42C24B724hsinnreiadobecom_--

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Henry Sinnreich wrote:
> >it's trying to stop SIP servers being used as DOS amplifiers.
>
> If you take this viewpoint, then the fewer servers, the less
> vulnerability?
>
> Henry

In this context, "SIP server" means both proxies and UAs (at least,
their UAS component).  Eliminating B2BUAs (which I think is what you
were getting at) sadly won't eliminate this attack.

Regards,

Michael

From cnmjbm@gmail.com  Fri Feb 20 08:41:21 2009
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From: "cow" <cnmjbm@gmail.com>
To: "Nils Ohlmeier" <lists@ohlmeier.org>
References: <001101c992f7$527fbb90$840010ac@panpipi> <499E70CB.3000004@ohlmeier.org>
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Subject: Re: [Sip] questions about usage of tel URI with a SIP UA
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Hi, Nils.

Thank you a lot for the comments.

Before I ask new categories of questions on sip-implementors list, may i 
elaborate my question and could you clarify your comments a little more ?

As i know of RFC 3455, the URIs returned in P-Associate-ID header of 
REGISTER 200 response are to be populated as Address-of-Record of the 
registered UA and are to be used in FROM headers of subsequent outward 
requests by the UA.

Let's take these *pseudo* SIP msgs for example:

=> UA 'a' registers itself via its outbound proxy with this REGISTER 
request:

        REGISTER sip:domain
        To: aor_of_a
        From: aor_of_a (or, aor_of_3rd-party)
        Contact: LOCATION_uri_of_a

=> Registrar returns 200:

        200 OK
        To: aor_of_a
        From: aor_of_a (or, aor_of_3rd-party)
        P-Associated-ID: paid1_of_a, paid2_of_a, ...
        (let's assume paid1_of_a='1-408-444-5555')

=> Later, UA 'b' calls UA 'b':

        INVITE tel:+1-408-444-5555
        To: tel:+1-408-444-5555
        From: aor_of_b

My question is: doesn't this INVITE go to IMS core and be resolved to

        INVITE LOCATION_uri_of_a
        To: tel:+1-408-444-5555
        From: aor_of_b

before it can arrive at UA 'a' ????

That is, in my knowledge, once after location is looked up and resolved by a 
SIP proxy, the INVITE Request-URI should no more be a tel uri, am i correct?

Please advise.

Thanks.

Cowby

----- Original Message ----- 
From: "Nils Ohlmeier" <lists@ohlmeier.org>
To: "Cowby" <cnmjbm@gmail.com>
Cc: <sip@ietf.org>
Sent: Friday, February 20, 2009 12:58 AM
Subject: Re: [Sip] questions about usage of tel URI with a SIP UA


> Hello,
>
> Am 20.02.2009 2:05 Uhr, schrieb Cowby:
>> Q1: is it possible that a SIP UA can receive incoming requests with tel 
>> uri
>> as the request-uri?
>
> In IMS yes.
>
>> Q2: if the answer to Q1 is yes, how does a SIP UA normally handle tel uri 
>> to
>> match the UA itself?
>
> Take a look at the P-Associated-URI header.
>
>> For example, if a UA has a mailto:"sip:cowby@cows.com' as its IMS public
>> identity and +1-408-555-6666 as it global phone number,
>>
>> Q1: is it possible that the UA's P-CSCF can route an INVITE
>> tel:+1-408-555-6666 request to the UA?
>
> Yes, see above.
>
>> Q2: how can the UA match itself when it is configured as cowby@cows.com 
>> but
>> know nothing of its phone number?
>
> See above as well.
>
> BTW such questions should be directed to the sip-implementors mailing list 
> instead (although I'm not sure if they discuss IMS as well).
>
> Best regards
>   Nils Ohlmeier 


From dean.willis@softarmor.com  Fri Feb 20 12:28:35 2009
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On Feb 20, 2009, at 9:53 AM, Michael Procter wrote:

> Henry Sinnreich wrote:
>>> it's trying to stop SIP servers being used as DOS amplifiers.
>>
>> If you take this viewpoint, then the fewer servers, the less
>> vulnerability?
>>
>> Henry
>
> In this context, "SIP server" means both proxies and UAs (at least,
> their UAS component).  Eliminating B2BUAs (which I think is what you
> were getting at) sadly won't eliminate this attack.
>

I believe Henry would also like to reduce proxies. I'm becoming more  
and more fond of this as a general design principle. Quite possibly,  
the only proxies we should have are ones that provide the outbound/ 
registrar/rendezvous function set. And that's just because we don't  
have those functions at a lower transport level. Other proxy functions  
(like application dispatch and delivery)  may be pointless exercises  
in futility. A working HIP-like transport layer could eliminate the  
outbound/registrar/rendezvous functions.

Most of the other "proxy" functions  we've discussed are probably  
better as UA functions.

Of course, that doesn't help with an attack mode that turns UAs into  
amplifiers . . .

--
Dean


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From: "Radha Krishna Saragadam (rsaragad)" <rsaragad@cisco.com>
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Subject: Re: [Sip] [Technical Errata Reported] RFC4028 (1687)
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Correction in "Corrected Text"

Section: 9

Original Text
-------------
The UAS MUST NOT increase the value of the Session-Expires header field.


Corrected Text
--------------
If the request doesn't indicate support for the session timer but
contains a session interval that is too small, the UAS cannot
usefully reject the request,as this would result in a call failure.
Rather, the UAS SHOULD increase the Session-Expires header field value
to be equal to its minimum interval in the response.

Regards
S.Radha krishna

-----Original Message-----
From: RFC Errata System [mailto:rfc-editor@rfc-editor.org]=20
Sent: Thursday, February 19, 2009 4:32 PM
To: srd@cisco.com; Jonathan Rosenberg (jdrosen);
jon.peterson@neustar.biz; Cullen Jennings (fluffy);
dean.willis@softarmor.com; drage@alcatel-lucent.com
Cc: Radha Krishna Saragadam (rsaragad); sip@ietf.org;
rfc-editor@rfc-editor.org
Subject: [Technical Errata Reported] RFC4028 (1687)


The following errata report has been submitted for RFC4028, "Session
Timers in the Session Initiation Protocol (SIP)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3D4028&eid=3D1687

--------------------------------------
Type: Technical
Reported by: Radha krishna Saragadam <rsaragad@cisco.com>

Section: 9

Original Text
-------------
The UAS MUST NOT increase the value of the Session-Expires header field.


Corrected Text
--------------
same as session 8.1
   If the request doesn't indicate support for the session timer but
   contains a session interval that is too small, the UAS cannot
   usefully reject the request, as this would result in a call failure.
   Rather, the UAS SHOULD insert a Min-SE header field containing its
   minimum interval.  If a Min-SE header field is already present, the
   UAS SHOULD increase (but MUST NOT decrease) the value to its
   minimum interval.  The UAS MUST then increase the Session-Expires
   header field value to be equal to the value in the Min-SE header
   field

Notes
-----
----- Forwarded Message ----
From: Radha krishna <krishna_srk2003@yahoo.com>
To: Brett Tate <brett@broadsoft.com>;
"sip-implementors@lists.cs.columbia.edu"
<sip-implementors@lists.cs.columbia.edu>
Sent: Thursday, February 19, 2009 10:56:31 AM
Subject: Re: [Sip-implementors] Sending 422

Thanks Brett,

      So I think same should be added for UAS

<Snip from RFC section 8.1>

  If the request doesn't indicate support for the session timer but
=20
contains a session interval that is too small, the proxy cannot
  usefully
reject the request, as this would result in a call failure.
  Rather, the
proxy SHOULD insert a Min-SE header field containing its
  minimum
interval.  If a Min-SE header field is already present, the
  proxy SHOULD
increase (but MUST NOT decrease) the value to its
  minimum interval.  The
proxy MUST then increase the Session-Expires
  header field value to be equal to the value in the Min-SE header
  field, as described above.
</Snip from RFC>


Regards
S.Radha krishna




________________________________
From: Brett Tate <brett@broadsoft.com>
To: Radha krishna <krishna_srk2003@yahoo.com>;
"sip-implementors@lists.cs.columbia.edu"
<sip-implementors@lists.cs.columbia.edu>
Sent: Wednesday, February 18, 2009 6:42:31 PM
Subject: RE: [Sip-implementors] Sending 422

It looks like Section 9 may have forgotten to indicate the behavior when
UAC timer support not indicated.  Section 8.1 allows a proxy to increase
the Session-Expires; I see no reason why the same cannot be done by the
UAS.

> -----Original Message-----
> From: sip-implementors-bounces@lists.cs.columbia.edu [mailto:sip-=20
> implementors-bounces@lists.cs.columbia.edu] On Behalf Of Radha krishna
> Sent: Tuesday, February 17, 2009 9:48 PM
> To: sip-implementors@lists.cs.columbia.edu
> Subject: [Sip-implementors] Sending 422
>
> Hi
>
>          Consider the following topology
>              UA1 ----- Call-stateful-proxy ------ UA2
>
>          UA1 does not support session timer, Make a call to UA2. Call-

> stateful-proxy adds Session-Expires:100 header and forwards to UA2.=20
> UA2 minimum session expires is 900. But in this case INVITE will not=20
> contain
> "support: timer". According section 9, UAS can reject with 422 only if

> there is a timer tag in supported header
>
> <Snip from RFC>
>
>    If an incoming request contains a Supported header field with a
value
>    'timer' and a Session Expires header field, the UAS MAY reject the
>    INVITE request with a 422 (Session Interval Too Small) response if
>    the session interval in the Session-Expires header field is smaller
>    than the minimum interval defined by the UAS' local policy.  When
>    sending the 422 response, the UAS MUST include a Min-SE header
field
>    with the value of its minimum interval.  This minimum interval MUST
>    NOT be lower than 90 seconds.
> </Snip from RFC>
>
>        Also UAS cannot increase the session expires duration <Snip=20
> from RFC> The UAS MUST
>    NOT increase the value of the Session-Expires header field.
> </Snip from RFC>
>
> What should be the behavior of UAS here?
>    1) accept the call with 100 seconds?
>    2) Increase the duration to 900 seconds while sending 200 Ok?
> Note: Session timer should not be turned-off
>
> Regards
> S.Radha krishna
>
>
>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@lists.cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors



    =20
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Sip-implementors mailing list
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Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please use
"Reply All" to discuss whether it should be verified or rejected. When a
decision is reached, the verifying party (IESG) can log in to change the
status and edit the report, if necessary.=20

--------------------------------------
RFC4028 (draft-ietf-sip-session-timer-15)
--------------------------------------
Title               : Session Timers in the Session Initiation Protocol
(SIP)
Publication Date    : April 2005
Author(s)           : S. Donovan, J. Rosenberg
Category            : PROPOSED STANDARD
Source              : Session Initiation Protocol
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From theo@crazygreek.co.uk  Fri Feb 20 13:05:19 2009
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Date: Fri, 20 Feb 2009 21:05:30 +0000
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
To: Dean Willis <dean.willis@softarmor.com>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, SIP WG <sip@ietf.org>, Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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On Fri, Feb 20, 2009 at 8:28 PM, Dean Willis <dean.willis@softarmor.com> wrote:

> Of course, that doesn't help with an attack mode that turns UAs into
> amplifiers . . .

Well, luckily UAs could more easily remain stateless when rejecting
responses than proxies, as they don't need to worry about things like
forking (be it parallel or serial - e.g due to next hop SRV
processing), so should be able to commonly respond to failure
responses statelessly.

This specific attack only arises when a there can be a invite server
transaction created directly by a UDP request that has not been
authenticated, and doesn't contain a "via cookie"

alas, every UA implementation i've played with to date has almost
always been stateful for even failure responses sent over UDP that
havn't required a large amount of processing (or any variance) to
reach it, except for some very low level induced responses like
malformed header fields.

 ~ Theo

From dworley@nortel.com  Fri Feb 20 14:23:55 2009
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From: "Dale Worley" <dworley@nortel.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Date: Fri, 20 Feb 2009 17:24:03 -0500
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Cc: SIP WG <sip@ietf.org>
Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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Some quasi-nits regarding the presentation, followed by more substantive
observations:

On page 3, you do an accounting for the level of amplification and total
22 responses for a spoofed INVITE.  But of course, if the spoofed call
can be directed to a UA that will answer (voicemail, anyone?), there can
be far more than 22 RTP packets generated.

On page 7, the process of resending a response that was initially
rejected with a 4XX is discussed.  This should be clarified by noting
that the resent request is a further fork of the original request, which
is much different than the resending operation that is done by a UAC in
response to a 401/407 response.  In particular, the CSeq is not
incremented.

In the discussion, the terms "the client" and "the server" are used,
which would lead people to think of SIP elements in the normal way.  But
in this case, each input or output address/port/transport triple of an
element functions separately:  A cookie used to send from a triple can
only be used to send from that triple, and similarly, the client must
cache known valid cookies for each destination triple separately.

The proposed syntax for cookie is described in the text as using "-" as
a separator, but the BNF uses ":".  "-" should be used, as ":" is
invalid in a "token".

The example shows a request being sent with a value-less cookie
parameter, though the BNF immediately above does not allow such a
parameter.  More importantly, such a value-less parameter does nothing
useful:  If the client will accept requests without cookies, a request
without the parameter will be accepted.  If the client will not accept
requests without cookies, a request without the parameter will elicit a
4XX response, which can be used to resend the request.

In regard to reusing the branch parameters of the initial request in the
resent request -- please do not do that.  Implementations already depend
on proper uniqueness of branch parameters.  In particular, a client that
indexes its transaction states by branch value may respond to the resent
request with the response that was given to the initial request, viz.,
4XX.

In regard to client address triples:  For security, the client must
generate a cookie that is valid for the address to which the response
will be sent.  And validating the cookie requires that the address from
which the request is sent matches the address within the cookie.
Together, these require "symmetric signaling", that responses always go
to the same address from which their requests came, or more exactly, the
address from which the resent request will come.  In any case, the
interaction with the SIP rules in section 18.2.2 need to be enunciated
more clearly:

      o  Otherwise (for unreliable unicast transports), if the top Via
         has a "received" parameter, the response MUST be sent to the
         address in the "received" parameter, using the port indicated
         in the "sent-by" value, or using port 5060 if none is specified
         explicitly.  If this fails, for example, elicits an ICMP "port
         unreachable" response, the procedures of Section 5 of [4]
         SHOULD be used to determine where to send the response.

      o  Otherwise, if it is not receiver-tagged, the response MUST be
         sent to the address indicated by the "sent-by" value, using the
         procedures in Section 5 of [4].

As Shamik notes, the cookie needs to be used not only to validate that
the request comes from the address contained in the cookie, but also to
validate the requested "sent-by" value, the address to which the
response to the later request is to be sent.

Actually, the cookie isn't needed to validate where the request comes
from, but only to validate where the response will be sent.  This allows
us to escape the requirement for symmetric signaling described above --
cookies are generated based on the *destination* of the 4XX, and are
validated against the *response destination address* not the *request
source address*.  This still has the required security property -- the
client can only obtain the cookie for a destination D if D cooperates in
capturing the cookie and providing it to the client.  This does render
unusable the clause "If this fails...", because the client has to know
absolutely what the response destination will be.

(Or can the requester provide multiple cookies for multiple possible
destinations?)

Of course, this mechanism only give benefit if servers start enforcing
cookies, which can't be done until clients implement them.  But the
mechanism can be phased in, in that client implementations can be
deployed first.

Dale



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Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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Hi Theo,

Am 20.02.2009 16:36 Uhr, schrieb Theo Zourzouvillys:
> On Fri, Feb 20, 2009 at 9:46 AM, Nils Ohlmeier<lists@ohlmeier.org>  wrote:
>> I think the whole "attack scenario" which is described in the draft works
>> only if the attacker uses the proxy/registrar of the victim as a relay.
>> Reason 1): because the attacker hopefully does not know which UDP port of
>> the victim is currently "open" to accept incoming UDP packets. And if their
>> is a NAT or firewall in between the UA it will hopefully only accept UDP
>> packets from the proxy/registrar. If the UA of the victim runs on a public
>> IP without any "protection" it might accept SIP requests from anywhere,
>> although I would call this a very bad UA implementation.
>
> A victim does not need to have an open UDP port;  An attack can easily
> be launched against any IP address, even a non SIP related one.

Yes, similar to the DNS server aplimifaction attacs which are going on 
right now. I think should you should spend an extra section on the 
scenario to seperate it from the pure SIP scenario.

> A device behind NAT on the end of a link additionally has no
> protection at an IP level, as the link would easily become saturated
> as a reflector starts sending failure response retransmits from a IST
> every time timerG fires.

Ok, I fear a shortage of you draft caused some confusion here:
IMHO you should differentiate between the attacker, the amplifier and 
the victim.
I was referring here to a UAC behind a NAT being the amplifier. You 
seemed to refer to the real victim here.

With this said I think it would be good if you seperate in your draft 
between UA aplimifers and proxy amplifiers, because they result in 
different traffic numbers if I'm not wrong.

Thus my original though was and still is, that it is most likely easier 
to mis-use any SIP proxy as amplifier. But a proxy will most likely 
"only" re-send any error response 11 times to the victim.
I still believe a UAC should be harder to mis-use as an amplifier (see 
original reasons 1 and 2).

>> Reason 2): when the UA of the victim registeres itself it provides a Contact
>> URI. I general I believe UA should only accept incoming requests from their
>> registrar IP and port which contains the Contact URI which they provided at
>> the registration. This prevents very easily that someone sends drive by
>> requests just with the AOR of the victim as R-URI to the UA of the victim.
>> If a registrar forwards requests to registered UAs with the AOR of the user
>> in the R-URI the registrar is broken in my opinion.
>
> as above - a victim does not need to be a SIP UA - it doesn't even
> need to talk SIP.  The attacker may well be trying to attack a non SIP
> entity, for example a web, DNS, or mail server.
>
> On the flip side, a UA that "only accept"s requests from it's
> registrar if of course good practice, but is only a step in stopping
> SPIT, not denial of service (which this draft tries to tackle), unless
> the "only accept" is limited by routability.  Note even NAT doesn't
> stop a denial of service attack being launched against the NAT
> gateway, causing it's connectivity to become saturated.

See above.
As a victim it makes for sure no difference under which criterias a SIP 
implementation (if it speeks SIP at all) accepts messages. But as an 
amplifier it might make a difference. E.g. if the UA rejects the spoofed 
INVITE with a 305 Use Proxy (although this reply is ambigious), the 
victim will receive only 11 replies within 32 seconds. But if the UA 
accepts the INVITE and starts to ring, and the user accepts... your 
worst case scenario comes into play.

> This draft is specifically trying to stopping a SIP server transaction
> being used as an amplifying reflector in a (distributed) denial of
> service attack.  The victim of the attack does not need to be a SIP
> endpoint (it doesn't even need to have any ports open!)
>
>> Now the question is of the proxy/registrar accepts requests from its users
>> from anywhere. I guess that is what you mean with no authentication, right?
>
> Yes - any SIP server (note a UAS or proxy are both servers for this)
> which allows an unauthenticated, untrusted user to create an invite
> server transaction (even for failure delivery) is vulnerable.

Agreed. Although a failed server transaction results in a lot less harm 
then an accepted call. Maybe some calculations about the amount of bytes 
would be helpfull here.

>> One bigger concern for the whole approach is: if the attacker knows about
>> this cookie extension he will for sure not put an empty cookie in the Via
>> header of the request. But how does the proxy of the victim handle requests
>> without the cookie extension?
>
> Any proxy that does wish to be used as an amplifier should reject
> requests without cookies with a stateless 4XX, or send a stateless
> 401/407 if knows it can authenticate it's user.
>
> Of course this is not going to be realistic from day 1, but this
> document is a start in finding solutions to some of the DOS problems
> we're currently susceptible to by putting a SIP server "on the net".
>
> Right now, there are a large number of SIP endpoints out there that
> will amplify a single UDP packet sent to them, and the initiator of
> the attack isn't even traceable!

Yep. Like with this DNS amplification attack here:
http://isc.sans.org/diary.html?storyid=5713

(And that only uses the amplification of bytes from a 1:1 
request:response transaction... SIP is worse.)

> (i'm starting to sound like a scaremonger, so i'll stop now :))
>
>> If the proxy of the victim does not care about backward compatibility (maybe
>> because attack previntion is more important), wouldn't the need a way to
>> tell the sender of the request that the cookie extension is required to
>> successfully relay requests? So wouldn't we need a new value for the
>> Required header, or would a new 4XX response already be enough.
>
> I did briefly consider an option tag but the problem is a SIP UA does
> not need to implement anything for it to potentially work - for
> example a proxy can implement it on behalf of a group of trusted users
> (for example, a corporate proxy or an outbound proxy that has
> authenticated the inbound request).
>
> There would additionally be no mechanism for recovery if the server
> refuses to process UDP requests without cookie support.  The 4XX would
> provide a code that allows a machine to understand that it failed due
> to cookies (if it knew about, but didn't implement the extension) -
> and could for example try upgrading to TCP, [D]TLS etc.  The reason
> phrase provides something descriptive for devices that don't
> understand it.  For those that don't, it'll fall back to treating it
> like a 400.

Ok, accepted.

Cheers
   Nils

From theo@crazygreek.co.uk  Fri Feb 20 15:34:49 2009
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Date: Fri, 20 Feb 2009 23:34:59 +0000
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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On Fri, Feb 20, 2009 at 10:24 PM, Dale Worley <dworley@nortel.com> wrote:

Hi Dale,

Thank you very much for your feedback.

I've integrated all your comments into my working -01 draft, with some
additional comments here (i'll reply to your other thoughts
separately):

> On page 3, you do an accounting for the level of amplification and total
> 22 responses for a spoofed INVITE. =A0But of course, if the spoofed call
> can be directed to a UA that will answer (voicemail, anyone?), there can
> be far more than 22 RTP packets generated.

absolutely! i'll elaborate on this more in -01, although i'm trying to
be careful not to imply that this particular draft is attempting to
solve RTP sink or in-dialog route/contact verification and only "drive
by shooting" you can do with UDP - i've started a draft that documents
that side of things (and a proposed solution) and will try to submit a
-00 over the weekend.

> More importantly, such a value-less parameter does nothing
> useful: =A0If the client will accept requests without cookies, a request
> without the parameter will be accepted. =A0If the client will not accept
> requests without cookies, a request without the parameter will elicit a
> 4XX response, which can be used to resend the request.

A server could behave differently when it receives a request without a
cookie in it, for example require authentication, redirect the user
statelessly to another transport, or something we've not thought of
just yet :-)

It's also useful for diagnostic purposes for deployment: how will we
know when adoption is so widespread it would be safe to require
support for it for a request to proceed?

> In regard to reusing the branch parameters of the initial request in the
> resent request -- please do not do that.

agreed - i'd already removed from my current rev.  It was about 4am
when i wrote that, and didn't think it through to well :-)

 ~ Theo

From theo@crazygreek.co.uk  Fri Feb 20 16:06:41 2009
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Hi Nils,

On Fri, Feb 20, 2009 at 11:22 PM, Nils Ohlmeier <lists@ohlmeier.org> wrote:

> Yes, similar to the DNS server aplimifaction attacs which are going on right
> now. I think should you should spend an extra section on the scenario to
> seperate it from the pure SIP scenario.
>
> [snip]
>
> With this said I think it would be good if you seperate in your draft
> between UA aplimifers and proxy amplifiers, because they result in different
> traffic numbers if I'm not wrong.

Will do.

> Agreed. Although a failed server transaction results in a lot less harm then
> an accepted call.

although "lot less harm" is still "way more harm than any other
commonly deployed publicly accessible UDP based service" :-)

> Maybe some calculations about the amount of bytes would be helpfull here.

sure.

 ~ Theo

-- 
Theo Zourzouvillys
Chief Technical Officer
VoIP.co.uk

From theo@crazygreek.co.uk  Fri Feb 20 16:44:48 2009
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Hi Dale,

On Fri, Feb 20, 2009 at 10:24 PM, Dale Worley <dworley@nortel.com> wrote:

> Actually, the cookie isn't needed to validate where the request comes
> from, but only to validate where the response will be sent. =A0This allow=
s
> us to escape the requirement for symmetric signaling described above --
> cookies are generated based on the *destination* of the 4XX, and are
> validated against the *response destination address* not the *request
> source address*. =A0This still has the required security property -- the
> client can only obtain the cookie for a destination D if D cooperates in
> capturing the cookie and providing it to the client. =A0This does render
> unusable the clause "If this fails...", because the client has to know
> absolutely what the response destination will be.

This actually seems like a far better way, althoguh the original
reasoning was to handle the (all be it pathological) case of different
stacks running on the same IP address but a different port as visible
by a proxy - perhaps for example due to (argh!) CGNAT e.g on most
mobile networks (at least in the UK) , or a serviced office which does
NAT to the same external IP for all it's hosts.

saying that, i'm not too fussed either way, and would appreciate any
thoughts on it...

> (Or can the requester provide multiple cookies for multiple possible
> destinations?)

I did consider this previously, but can'y see any way of being able to
handle the case where sent-by contains a host name which resolves to
multiple SRV records with same priority, as the response could go to
any of them (and each would need to be validated and get it's own
token) before passing it to the client to include in the request.
cookies would also need to have a minimum lifetime, so that a server
knows how long those it's generated are valid, and the whole thing
becomes horribly complex... :-)

my thinking was along the lines of because the response is going to be
UDP, a server that needs the redundancy can use anycast on the single
IP / port to ensure to gets the response, even if a given proxy (or
even data centre) dies.

 ~ theo

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From: Henry Sinnreich <hsinnrei@adobe.com>
To: Theo Zourzouvillys <theo@crazygreek.co.uk>, Dean Willis <dean.willis@softarmor.com>
Date: Fri, 20 Feb 2009 18:14:20 -0800
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--_000_C5C4BF9CB77Dhsinnreiadobecom_
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Please correct me, but these and previous other discussions on related topi=
cs, such as forking, early media and race conditions seem to confirm the fa=
ct that no formal and reliable routing protocol has been developed as yet t=
o route between a larger number (more than 2 in the trapezoid model) of SIP=
 proxies and various (Oh so many) other feature and media servers.

If there were a reliable routing protocol between all the SIP network eleme=
nts these discussions would not take place for so long and there would be j=
ust one RFC describing something like "The SIP Network Routing Protocol". A=
nd some open source implementation to prove it.

As is, every SIP network must be manually engineered, for every configurati=
on instance and adding yet another new "network service" requires starting =
all over again.

Is this view embarrassingly na=EFve?
It is only meant to defend the view SIP should be used just as a rendezvous=
 and session setup protocol.

Henry


On 2/20/09 3:05 PM, "Theo Zourzouvillys" <theo@crazygreek.co.uk> wrote:

On Fri, Feb 20, 2009 at 8:28 PM, Dean Willis <dean.willis@softarmor.com> wr=
ote:

> Of course, that doesn't help with an attack mode that turns UAs into
> amplifiers . . .

Well, luckily UAs could more easily remain stateless when rejecting
responses than proxies, as they don't need to worry about things like
forking (be it parallel or serial - e.g due to next hop SRV
processing), so should be able to commonly respond to failure
responses statelessly.

This specific attack only arises when a there can be a invite server
transaction created directly by a UDP request that has not been
authenticated, and doesn't contain a "via cookie"

alas, every UA implementation i've played with to date has almost
always been stateful for even failure responses sent over UDP that
havn't required a large amount of processing (or any variance) to
reach it, except for some very low level induced responses like
malformed header fields.

 ~ Theo


--_000_C5C4BF9CB77Dhsinnreiadobecom_
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<HTML>
<HEAD>
<TITLE>Re: [Sip] draft-zourzouvillys-sip-via-cookie-00</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
14pt'>Please correct me, but these and previous other discussions on relate=
d topics, such as forking, early media and race conditions seem to confirm =
the fact that no formal and reliable routing protocol has been developed as=
 yet to route between a larger number (more than 2 in the trapezoid model) =
of SIP proxies and various (Oh so many) other feature and media servers.<BR=
>
<BR>
If there were a reliable routing protocol between all the SIP network eleme=
nts these discussions would not take place for so long and there would be j=
ust <I>one RFC</I> describing something like &#8220;The SIP Network Routing=
 Protocol&#8221;. And some open source implementation to prove it.<BR>
<BR>
As is, every SIP network must be manually engineered, for every configurati=
on instance and adding yet another new &#8220;network service&#8221; requir=
es starting all over again.<BR>
<BR>
Is this view embarrassingly na&iuml;ve? <BR>
It is only meant to defend the view SIP should be used just as a rendezvous=
 and session setup protocol.<BR>
<BR>
Henry <BR>
<BR>
<BR>
On 2/20/09 3:05 PM, &quot;Theo Zourzouvillys&quot; &lt;<a href=3D"theo@craz=
ygreek.co.uk">theo@crazygreek.co.uk</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:14pt'>On Fri, Feb 20, 2009 at 8:28 PM, Dean Willi=
s &lt;<a href=3D"dean.willis@softarmor.com">dean.willis@softarmor.com</a>&g=
t; wrote:<BR>
<BR>
&gt; Of course, that doesn't help with an attack mode that turns UAs into<B=
R>
&gt; amplifiers . . .<BR>
<BR>
Well, luckily UAs could more easily remain stateless when rejecting<BR>
responses than proxies, as they don't need to worry about things like<BR>
forking (be it parallel or serial - e.g due to next hop SRV<BR>
processing), so should be able to commonly respond to failure<BR>
responses statelessly.<BR>
<BR>
This specific attack only arises when a there can be a invite server<BR>
transaction created directly by a UDP request that has not been<BR>
authenticated, and doesn't contain a &quot;via cookie&quot;<BR>
<BR>
alas, every UA implementation i've played with to date has almost<BR>
always been stateful for even failure responses sent over UDP that<BR>
havn't required a large amount of processing (or any variance) to<BR>
reach it, except for some very low level induced responses like<BR>
malformed header fields.<BR>
<BR>
&nbsp;~ Theo<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C5C4BF9CB77Dhsinnreiadobecom_--

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From theo@crazygreek.co.uk  Sat Feb 21 23:32:41 2009
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Message-ID: <167dfb9b0902212332t5ae3fdbcma49e89bc703358f2@mail.gmail.com>
From: Theo Zourzouvillys <theo@voip.co.uk>
To: SIP WG <sip@ietf.org>
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Subject: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
X-BeenThere: sip@ietf.org
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Hi,

i've just submitted a -01 based on the feedback i've received.

While we've all known about the issue forever, i've never sat down to
work out how bad it really is.

The results are pretty scary: "best" case scenario is a 1:10
amplification, and "worst" i can easily get a 1:350 amplification by
writing a quick script.

Additionally, out of every single vendor's implementation i looked at,
i've not yet found 1 that isn't vulnerable to being used in an attack:
phones, proxies, and SBCs can all be made to participate.

i'm surprised we've not seen this in the wild yet.  I've been recently
scanning for publicly accessible SIP servers, and out of a /8 (16
million hosts) that i've got through so far, there have been 72,725
SIP servers that are responding on port 5060 UDP.  that's a lot of
amplifiers that can be used in this attack: and that's only 1/100th of
the currently allocated RIR IPv4 address space.

 ~ Theo

From theo@crazygreek.co.uk  Sat Feb 21 23:48:53 2009
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References: <167dfb9b0902201305k3aa3d271k59f770218be98283@mail.gmail.com> <C5C4BF9C.B77D%hsinnrei@adobe.com>
Date: Sun, 22 Feb 2009 07:49:08 +0000
Message-ID: <167dfb9b0902212349r653cc5cta79c7fd65ae4995c@mail.gmail.com>
From: Theo Zourzouvillys <theo@crazygreek.co.uk>
To: Henry Sinnreich <hsinnrei@adobe.com>
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Cc: Nils Ohlmeier <lists@ohlmeier.org>, SIP WG <sip@ietf.org>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie-00
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On Sat, Feb 21, 2009 at 2:14 AM, Henry Sinnreich <hsinnrei@adobe.com> wrote=
:

> Is this view embarrassingly na=EFve?

Well, i'd prefer to stay out of a discussion on anything other than
getting the issue at hand fixed, as it's a major problem *right* *now*
for operators who have no choice but to implement SIP/2.0 at it is on
the public internet.

On the other hand, i'll happily engage in any discussion about
anything at all in SF :-)

(on that note, i'm going to be around SF/mountain view the whole of
the week before the IETF if anyone else is and wants to meet up for
drinks or food and general SIP chat - contact me offlist).

 ~ Theo

From HKaplan@acmepacket.com  Sun Feb 22 08:21:11 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@voip.co.uk>, SIP WG <sip@ietf.org>
Date: Sun, 22 Feb 2009 11:19:37 -0500
Thread-Topic: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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Subject: Re: [Sip] SIP amplification vulnerability -01	(draft-zourzouvillys-sip-via-cookie-01)
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of The=
o
> Zourzouvillys
> Sent: Sunday, February 22, 2009 2:33 AM
>
> While we've all known about the issue forever, i've never sat down to
> work out how bad it really is.
> The results are pretty scary: "best" case scenario is a 1:10
> amplification, and "worst" i can easily get a 1:350 amplification by
> writing a quick script.
>
> Additionally, out of every single vendor's implementation i looked at,
> i've not yet found 1 that isn't vulnerable to being used in an attack:
> phones, proxies, and SBCs can all be made to participate.

I'm not debating your draft's premise, but that last statement is highly de=
pendent on the system and deployment's configuration.  For the remainder of=
 this I'll use the term "SBC", but as far as I know proxies can do this too=
.

SBCs can be and almost always are configured to only accept requests from r=
egistered UA's.  So let's say you guessed one or many of them correctly; bo=
th their AoR's and guessed their full contact URI's (including username) an=
d Via/Layer-3 IP:ports as well.  In typical deployments the request will be=
 challenged, so the amplification is the 100 and 401/407 challenges, not a =
full call's worth of 18x/200 and RTP.  (And in practice there are some tric=
ks the SBC can do to reduce actual amplification to 1:1, but it's rare to e=
nable it - if amplification attacks are actually encountered, then this may=
 change)

So the next issue is attack volume, and again it depends.  SBCs can and oft=
en do throttle the number of requests they'll accept in any time period fro=
m any given UA.  So they won't flood the target UA with bogus responses, be=
cause they won't accept that many spoofed requests from it to begin with - =
for some vendors they have hardware that will silently drop the excess.

What will sometimes happen is the SBC will dynamically black-list a floodin=
g UA, so that if you successfully spoof one you can knock it out of service=
 anyway, for a period of time.  Of course this triggers traps to notify the=
 admins to look into it, but at least temporarily the damage is done. But t=
hat's by design - if you're successfully spoofing another UA, then the best=
 an SBC can do is limit the impact of that to affect only the spoofed UA an=
d not impact other users.

The harder problem is SIP trunks, a la IP-PBX's.  If they aren't a register=
ing type of PBX, then their addresses are usually provisioned in the SBC's =
in terms of ACLs; both layer-3 ACLs and layer-7, such that you'd still have=
 to spoof more than just IP:port.  But it's a harder problem because it's e=
asier to guess that stuff for an Enterprise than from a subscriber, and bec=
ause they often aren't challenge-able.  They're still usually throttled; an=
d in theory potentially blacklisted, though that's less common to enable fo=
r PBX connections.  The only good side of it is the bigger the Enterprise, =
the more likely the SIP trunk is over a VPN of some type, where the spoofed=
 requests would have to come in on that VPN to be processed.

BTW, we do recommend to providers that they employ uRPF on their edge route=
rs, and some can and do.  Others don't have that ability or don't own the w=
hole network, or don't want to take the performance impact of doing it on t=
heir routers. (because many routers take a hit for uRPF)

-hadriel

From theo@crazygreek.co.uk  Sun Feb 22 12:32:22 2009
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Subject: Re: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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On Sun, Feb 22, 2009 at 4:19 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wr=
ote:

> SBCs can be and almost always are configured to only accept requests from=
 registered UA's.

What about an SBC that is configured at the edge of a network, for
example the thing that is the target of the URIs placed into ENUM?  Or
service providers who want to accept calls from "the internet" through
an SBC? These are vulnerable, are they not?

... i'm not worried about an amplification risk with a user that can
be authenticaticated: the 401/407 can be sent statelessly, meaning
only a very small amplification ratio and only 1:1 packets. (although
note that most UAs and proxies i tested did not currently send them
statelessly and instead created a IST for them along with them and
fire timer G!) - i imagine a well designed SBC would perform
differently, although i didn't happen to have a well designed SBC from
a certain vendor available for testing, so couldn't say :-)

>=A0In typical deployments the request will be challenged, so the amplifica=
tion is the 100 and 401/407 challenges

are there ever any reasons for an SBC to not send the 401/407
statelessly, and even send a 100?  A 1:2 attack is still bad, albeit a
far cry from the pathological case.

> not a full call's worth of 18x/200 and RTP.

bear in mind the via-cookie doesn't stop the in-dialog/rtp targeting,
but it does mean that it's traceable, and thus "manageable" once a
solution like via-cookie is in place. I think work needs to be done on
that issue too, but it's no where near as risky as this issue is.

> SBCs can and often do throttle the number of requests they'll accept in a=
ny time period from any given UA. =A0So they won't flood the target UA with=
 bogus responses, because they won't accept that many spoofed requests from=
 it to begin with - for some vendors they have hardware that will silently =
drop the excess.

Having a single SIP element doing rate limiting doesn't matter though
when there are 7.2 million SIP devices on the net [1].  If each one
will allow a 1:10 amplification, that's all a botnet needs to make it
10 times more valuable.  Hell, even a 1:2 amplification is worth it
for them, especially as it's reflected.

> What will sometimes happen is the SBC will dynamically black-list a flood=
ing UA, so that if you successfully spoof one you can knock it out of servi=
ce anyway, for a period of time. =A0Of course this triggers traps to notify=
 the admins to look into it, but at least temporarily the damage is done. B=
ut that's by design - if you're successfully spoofing another UA, then the =
best an SBC can do is limit the impact of that to affect only the spoofed U=
A and not impact other users.

... cookies solves this particular attack though, doesn't it?
(although authentication does as well in this scenario due to a
nonce?), in the same way a syncookie does for the SYN flooding
problems of decades gone by.

> BTW, we do recommend to providers that they employ uRPF on their edge rou=
ters, and some can and do.

absolutely.  over time we've got to the point where *most* UK SPs
worth their salt are doing RPF, but ....

>=A0Others don't have that ability or don't own the whole network, or don't=
 want to take the performance impact of doing it on their routers. (because=
 many routers take a hit for uRPF)

It's partially loosing battle trying to get service providers to do it
(~6 years years ago i went on a rampage to try and get UK ISPs to do
it - some did, most didn't.  thankfully most have since then, though),
and while it is important (it makes bot nets lifes harder) in the
global scheme of  things, as you say many don't - and it only takes
one to have a problem.  Even if everyone did RPF at the access layer,
it doesn't stop attackers.  They just become service providers
themself (this is what spammers have done as thigns got more difficult
for them).  You don't even need to inject a prefix into global routing
tables: simply transmitting the packets is all it takes, and won't
even be traceable.

There are bunches of issues at IXs as well: there is no way ports can
be L3 filtered there currently.  The SIDR WG is trying to find
solutions to some this, but it's a long way off becoming reality imo.

 ~ Theo

1 - completely random guess from my (72k * 100).

From fluffy@cisco.com  Sun Feb 22 13:43:07 2009
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Cc: sip@ietf.org, Keith Drage <drage@alcatel-lucent.com>, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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I'm looking at this and it seems to me this is not yet a WG draft - A  
WGLC might be a bit premature.


On Feb 19, 2009, at 4:12 AM, Christer Holmberg wrote:

>
> Hi,
>
> I've submitted a new version of draft-ietf-sip-199. I have done  
> changes and updates based on feedback and discussions (please let me  
> know if you think I've forgotten something), and I've also added  
> text to the security chapter.
>
> I propose that we start the WGLC process for the draft.
>
> The draft can also be found at:
>
> http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt
>
> Regards,
>
> Christer
>
> _______________________________________________
> Sip mailing list  https://www.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 brett@broadsoft.com  Sun Feb 22 14:41:29 2009
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To: Cullen Jennings <fluffy@cisco.com>
Date: Sun, 22 Feb 2009 14:41:44 -0800
Thread-Topic: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Cullen,

What do you mean by "not yet a WG draft"?  Didn't it become the WG draft Ma=
y 2008 to address the WG chartered item: "Termination of early dialog prior=
 to final response"? =20

The status within the charter appears to indicate that the WGLC is already =
"Done". :)=20

Other than maybe delaying until there are more implementations, is there a =
reason to delay the WGLC?  I doubt delaying the WGLC would get any of the n=
on interested participants to provide comments; however a WGLC might finall=
y get them to send comments other than indicate that they are not intereste=
d in completing the draft unless it also solves HERFP.

Thanks,
Brett

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Cullen Jennings
> Sent: Sunday, February 22, 2009 4:43 PM
> To: Christer Holmberg
> Cc: sip@ietf.org; Keith Drage; Dean Willis
> Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
>=20
>=20
> I'm looking at this and it seems to me this is not yet a WG draft - A
> WGLC might be a bit premature.
>=20
>=20
> On Feb 19, 2009, at 4:12 AM, Christer Holmberg wrote:
>=20
> >
> > Hi,
> >
> > I've submitted a new version of draft-ietf-sip-199. I have done
> > changes and updates based on feedback and discussions (please let me
> > know if you think I've forgotten something), and I've also added
> > text to the security chapter.
> >
> > I propose that we start the WGLC process for the draft.
> >
> > The draft can also be found at:
> >
> > http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt
> >
> > Regards,
> >
> > Christer


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Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] New version (-05) of the 199 draft - ready for WGLC?
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Brett, you are right - I just totally messed this one up and missed it  
when I went looking. My apologies - I was just trying to clean up too  
many emails too quickly. Please ignore my previous email and  finish  
this WG draft really soon :-)

Sorry, Cullen


On Feb 22, 2009, at 3:41 PM, Brett Tate wrote:

> Cullen,
>
> What do you mean by "not yet a WG draft"?  Didn't it become the WG  
> draft May 2008 to address the WG chartered item: "Termination of  
> early dialog prior to final response"?
>
> The status within the charter appears to indicate that the WGLC is  
> already "Done". :)
>
> Other than maybe delaying until there are more implementations, is  
> there a reason to delay the WGLC?  I doubt delaying the WGLC would  
> get any of the non interested participants to provide comments;  
> however a WGLC might finally get them to send comments other than  
> indicate that they are not interested in completing the draft unless  
> it also solves HERFP.
>
> Thanks,
> Brett
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Cullen Jennings
>> Sent: Sunday, February 22, 2009 4:43 PM
>> To: Christer Holmberg
>> Cc: sip@ietf.org; Keith Drage; Dean Willis
>> Subject: Re: [Sip] New version (-05) of the 199 draft - ready for  
>> WGLC?
>>
>>
>> I'm looking at this and it seems to me this is not yet a WG draft - A
>> WGLC might be a bit premature.
>>
>>
>> On Feb 19, 2009, at 4:12 AM, Christer Holmberg wrote:
>>
>>>
>>> Hi,
>>>
>>> I've submitted a new version of draft-ietf-sip-199. I have done
>>> changes and updates based on feedback and discussions (please let me
>>> know if you think I've forgotten something), and I've also added
>>> text to the security chapter.
>>>
>>> I propose that we start the WGLC process for the draft.
>>>
>>> The draft can also be found at:
>>>
>>> http://users.piuha.net/cholmber/drafts/draft-ietf-sip-199-05.txt
>>>
>>> Regards,
>>>
>>> Christer
>


From HKaplan@acmepacket.com  Sun Feb 22 15:56:44 2009
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Theo Zourzouvillys <theo@voip.co.uk>
Date: Sun, 22 Feb 2009 18:55:13 -0500
Thread-Topic: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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Subject: Re: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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> -----Original Message-----
> From: theo@crazygreek.co.uk [mailto:theo@crazygreek.co.uk] On Behalf Of
> Theo Zourzouvillys
> Sent: Sunday, February 22, 2009 3:33 PM
>
> On Sun, Feb 22, 2009 at 4:19 PM, Hadriel Kaplan <HKaplan@acmepacket.com>
> wrote:
>
> > SBCs can be and almost always are configured to only accept requests
> from registered UA's.
>
> What about an SBC that is configured at the edge of a network, for
> example the thing that is the target of the URIs placed into ENUM?

Yup, these are essentially the same as IP-PBX trunks.  I don't know of many=
 that allow requests in from truly unknown sources, though - they usually k=
now the remote devices in advance, even when ENUM is used - so if they want=
ed to they could use IPSec or TLS if this became a problem. (some do today,=
 but many don't - it seems many peer over private connections anyway)


> Or
> service providers who want to accept calls from "the internet" through
> an SBC? These are vulnerable, are they not?

Yup, but for those we strongly recommend IPsec, TLS, or even just TCP, wher=
e at least syncookies stops the trivial stuff.  Essentially what you're pro=
posing seems to me similar in concept to syncookies, but at the next layer =
up.


> are there ever any reasons for an SBC to not send the 401/407
> statelessly, and even send a 100?  A 1:2 attack is still bad, albeit a
> far cry from the pathological case.

Nope there's no reason I know of, but it's rare to do that.  Usually it's t=
he registrar or some core device which sends the 401/407, so the SBC sends =
a 100 to stop the UA from retransmitting.  I have tried to convince people =
to let the SBC do it, but it hasn't gotten that much interest. (despite the=
 fact it has more benefits than just this one issue)


> Having a single SIP element doing rate limiting doesn't matter though
> when there are 7.2 million SIP devices on the net [1].  If each one
> will allow a 1:10 amplification, that's all a botnet needs to make it
> 10 times more valuable.  Hell, even a 1:2 amplification is worth it
> for them, especially as it's reflected.

Your point is that botnets can take advantage of the SBC's to amplify their=
 attacks on other UA's, right?  But to do that they have to know the URI's/=
IP:port's of the other UA's they want to attack indirectly, and at that poi=
nt why bother attacking via the SBC - just attack the UA's directly, spoofi=
ng other UA's, or spoofing the SBC's, or even not spoofing.  I mean it's a =
botnet, so you have enough resources to cause a resource exhaustion attack =
on any UA's, and who cares if they can trace it?


> ... cookies solves this particular attack though, doesn't it?
> (although authentication does as well in this scenario due to a
> nonce?), in the same way a syncookie does for the SYN flooding
> problems of decades gone by.

Yup, either cookie or auth would solve it, but my guess is we'd still want =
to blacklist if it's more than a short-lived event.  That's what happens no=
w: you get some leeway for a short while, but eventually you get kicked out=
 for a longer period of time, because as far as we know you've got a broken=
 UA or someone's learned your info, and we can't let you affect the overall=
 network.

 -hadriel

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A new version of I-D, draft-worley-references-02.txt has been successfuly submitted by Dale Worley and posted to the IETF repository.

Filename:	 draft-worley-references
Revision:	 02
Title:		 The References Header for SIP
Creation_date:	 2009-02-22
WG ID:		 Independent Submission
Number_of_pages: 22

Abstract:
This document defines a SIP extension header, References, to be used
within SIP messages to signify that the message (and the dialog
containing it) is related to one or more other dialogs.  It is
expected to be used largely for diagnostic purposes.
                                                                                  


The IETF Secretariat.



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From drage@alcatel-lucent.com  Sun Feb 22 23:29:55 2009
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Cc: "sip@ietf.org" <sip@ietf.org>, "draft-ietf-sip-ua-privacy@tools.ietf.org" <draft-ietf-sip-ua-privacy@tools.ietf.org>
Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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The issue is that while the user may be wanting to use a public GRUU, they =
may not wish to announce their IMEI. Normally, no matter what privacy, IMEI=
s are kept secret in 3GPP.

You'll find procedures in the registration proxy in IMS to therefore obfusc=
ate the IMEI when creating a public GRUU. And in 3GPP this was not directly=
 an invention of the protocol group - part of this came from the 3GPP secur=
ity group.

regards

Keith=20

> -----Original Message-----
> From: Shida Schubert [mailto:shida@ntt-at.com]=20
> Sent: Thursday, February 19, 2009 7:53 AM
> To: DRAGE, Keith (Keith)
> Cc: Michael Procter;=20
> draft-ietf-sip-ua-privacy@tools.ietf.org; sip@ietf.org
> Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
>=20
>=20
> Keith;
>=20
>   I just looked at outbound and gruu again, and AFAICT=20
> temp-gruu doesn't carry the instance-id unlike the pub-gruu=20
> which does.
>=20
>   With outbound the instance-id is only present in the=20
> REGISTER request when you are establishing the flow, and if=20
> temp-gruu is used, the only time instance-id will be visible=20
> when GRUU is used is also in the REGISTER request.
>=20
>   I may be blind but I can't seem to see the problem.
>=20
>   Regards
>    Shida
>=20
> On 18-Feb-09, at 10:53 AM, DRAGE, Keith (Keith) wrote:
>=20
> > There is a similar issue with the instance-id as used by=20
> outbound and=20
> > GRUU as well. In a mobile, this uses the IMEI which is not=20
> necessarily=20
> > meant to be revealed.
> >
> > 3GPP obfuscate the IMEI on generation of the GRUU.
> >
> > regards
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]=20
> On Behalf Of=20
> >> Michael Procter
> >> Sent: Tuesday, February 17, 2009 8:51 AM
> >> To: draft-ietf-sip-ua-privacy@tools.ietf.org; sip@ietf.org
> >> Subject: [Sip] Comment on sip-ua-privacy-05.txt
> >>
> >> Just a minor point:  Is it worth adding (either in section
> >> 4.1 or 6) that a temp-gruu might not be as anonymous as you might=20
> >> hope?  An observer using RFC 3680 (reg-event) with gruu extensions=20
> >> would be able to correlate temp-gruus with AoRs and=20
> contacts, should=20
> >> they be so authorised.
> >>
> >> There is some text in RFC 3680 warning of the risks of=20
> reg-event, but=20
> >> that is probably of more direct interest to registrar authors.  A=20
> >> reminder of the risk in this document might highlight it for UA=20
> >> authors, so that they can consider the wider implications.
> >>
> >> Best regards,
> >>
> >> Michael
> >> _______________________________________________
> >> Sip mailing list  https://www.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
> >>
> > _______________________________________________
> > Sip mailing list  https://www.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
> =

From theo@crazygreek.co.uk  Mon Feb 23 02:49:00 2009
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Subject: Re: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
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On Sun, Feb 22, 2009 at 11:55 PM, Hadriel Kaplan <HKaplan@acmepacket.com> w=
rote:

> Yup, these are essentially the same as IP-PBX trunks. =A0I don't know of =
many that allow requests in from truly unknown sources, though - they usual=
ly know the remote devices in advance, even when ENUM is used - so if they =
wanted to they could use IPSec or TLS if this became a problem. (some do to=
day, but many don't - it seems many peer over private connections anyway)

I believe (hope!) that these are going to get more common! ... the
driving reason behind me doing this was running through security
issues for deployment of public gateways for the 4.4.e164.arpa go-live
date  on March 11th (at *last*).  While i'd love the world to change
to TCP/TLS overnight, i just don't see it happening ... just "cause
UDP can be used for attacking 3rd parties"

Even if SBCs may not be commonly deployed in that way, there are
certainly millions of SIP devices that are.  Think all the asterisk
machines hosted on public IPs, for example.

> Essentially what you're proposing seems to me similar in concept to synco=
okies, but at the next layer up.

yeah, it is - in fact probably why i subconsciously called them via cookies=
 :-)

> Your point is that botnets can take advantage of the SBC's to amplify the=
ir attacks on other UA's, right?

No!  Attacking UAs isn't really that much of a worry (as 401/407 can
"fix" that, especially as UDP scales so well).  It's that you can use
SBCs to amplify attacks on anything you like.

Botnets can take advantages of SIP elements to attack anything not
just UAs.  "anything" could be a DNS server, mail server, web server,
or anything with routable IP just due to sheer number of PPS and
throughput saturating links.

> I mean it's a botnet, so you have enough resources to cause a resource ex=
haustion attack on any UA's, and who cares if they can trace it?

Because it's another layer of indirection as well as amplification.
Indirection makes traceability harder, which means longer before
machines initiating attacks get discovered.  Amplification means
networks hosting the zombies are less likely to notice initially. so
it's a double fail for us :-)

> Yup, either cookie or auth would solve it, but my guess is we'd still wan=
t to blacklist if it's more than a short-lived event. =A0That's what happen=
s now: you get some leeway for a short while, but eventually you get kicked=
 out for a longer period of time, because as far as we know you've got a br=
oken UA or someone's learned your info, and we can't let you affect the ove=
rall network.

Lets say i have a 2mbit link at home.  Given the 1:350 example, I
could theoretically generate a 700mbit attack against anything i
wanted.

In the case of a botnet, they can use 4000 machines sending out a
single 500 byte INVITE every second, which would result in the same
thing.

 ~ Theo

From eric.wangxr@gmail.com  Mon Feb 23 07:18:00 2009
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From: Eric wang <eric.wangxr@gmail.com>
To: christer.holmberg@ericsson.com
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Subject: [Sip] Questions about "draft-ietf-sip-199-05"
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Hi Christer,
    Glad to see that the draft starts its WGLC process.It's really useful.

    I still have some difficult in using IETF and cannot find the whole
sorted comments about this draft, so i have some doubt about this draft.

   1.   "If a forking proxy receives a reliably sent 199 response for a

   dialog, for which the proxy has previously generated and sent a 199

   response, the proxy MUST forward the 199 response."
     Does it describe the case below? Although P1 have sent a 199 response,
P1 havs to  forword the send reliably 199 too.Or  is the  first 199 mistaked
for 18x?

          UAC               P1                  UAS_2
       --- INVITE ------>
                         --- INVITE (leg 2) ->
       <-- 199(leg 2) --
                         <-- 199 (leg 2) -----
       <-- 199(leg 2) --

     2.  "10.  Usage with 100rel

   When a 199 Early Dialog Terminated provisional response is sent by a

   UAS, since the provisional response is only used for information

   purpose, the UAS SHOULD send it unreliably even if the 100rel option

   tag [RFC3262] is present in the Require header of the associated

   request."



   I have seen a comment on this question,but still not understood about it
.

   If the INVITE has a Require tag "Require: 100rel",does the UAS still use

   unreliable 199 response?



   If 199 is reliable, there is one more advantage. If the 199 arrives
before

   the first 18x response, UAC can discard the first 199 and process it
until

   UAC receives the first 18x response that has the same to-tag as 199, as

   the reliable 199 should be re-transmited until  received PRACK.







   ps:



   Can anyone show me how to sort all comments of a topic by time?





   Regards,



   Eric.wang

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<div>Hi Christer,</div>
<div>&nbsp;&nbsp;&nbsp; Glad to see that the draft starts its&nbsp;WGLC pro=
cess.It&#39;s really useful.</div>
<div>&nbsp; </div>
<div>&nbsp;&nbsp;&nbsp; I still have some difficult in using IETF and canno=
t find the whole sorted comments about this draft, so i have some doubt abo=
ut this draft.</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; 1.&nbsp;&nbsp;&nbsp;<font color=3D"#3333ff">&quot;If a fo=
rking proxy receives a reliably sent 199 response for a </font></div>
<div>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#3333ff"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>dialog, for which the proxy has previously generated and sent a 199<=
/font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#3333ff"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>response, the proxy MUST forward the 199 response.&quot;</font></p><=
/div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Does it describe the case below?&nbsp;Al=
though P1 have sent a 199 response, P1 havs to&nbsp; forword the send relia=
bly 199 too.Or&nbsp; is the&nbsp; first 199 mistaked for 18x?</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp; UAC&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; P1&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; UAS_2&nbsp;&nbsp;&nbsp; </div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE ------&gt;<br>&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE (l=
eg 2) -&gt;</div>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199(leg 2) --<br><font col=
or=3D"#ff0000">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp; &lt;-- 199 (leg 2) -----<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;-- 199(leg 2) --</font></div>
<div><font color=3D"#ff0000"></font>&nbsp;</div>
<div><font color=3D"#ff0000">&nbsp;<font color=3D"#000000">&nbsp;&nbsp;&nbs=
p; 2.</font>&nbsp; <font color=3D"#000099">&quot;10.<span style=3D"mso-spac=
erun: yes">&nbsp; </span>Usage with 100rel</font>=20
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000099"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; </span></font><=
span style=3D"COLOR: #993300"><font color=3D"#000099">When a 199 Early Dial=
og Terminated provisional response is sent by a</font></span></p>

<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#000099"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>UAS, since the provisional response is only used for information</fo=
nt></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#000099"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>purpose, the UAS SHOULD send it unreliably even if the 100rel option=
</font></p>

<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#000099"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>tag [RFC3262] is present in the Require header of the associated</fo=
nt></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; COLOR: #993300; FONT-FAMILY: SimS=
un"><font color=3D"#000099"><span style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</span>request.&quot;</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;&nbsp;=
 </p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;<font =
color=3D"#000000">&nbsp;&nbsp;I have seen a comment on this question,but st=
ill not understood about it .</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; If the INVITE has a Require tag &quot;Require: 10=
0rel&quot;,does the UAS still use</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp;&nbsp;unreliable 199 response?</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000"></font>&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; If 199 is reliable, there is one more advantage. =
If the 199 arrives before </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; the first 18x response, UAC can discard the first=
 199 and process it until </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; UAC receives the first 18x response that has the =
same to-tag as 199, as </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; the reliable 199&nbsp;should&nbsp;be re-transmite=
d until&nbsp; received PRACK.</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000"></font>&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; ps:</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000"></font>&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; Can anyone show me how to&nbsp;sort all comments =
of a&nbsp;topic by time?</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000"></font>&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; </font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; Regards,</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000"></font>&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun"><font color=
=3D"#000000">&nbsp;&nbsp; Eric.wang</font></p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;&nbsp;=
 &nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;</p>
<p style=3D"FONT-SIZE: 10pt; MARGIN: 0in; FONT-FAMILY: SimSun">&nbsp;</p></=
font></div>

--000e0cd2bd52b2610b0463977d94--

From dean.willis@softarmor.com  Mon Feb 23 07:43:14 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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On Feb 23, 2009, at 1:30 AM, DRAGE, Keith (Keith) wrote:

> The issue is that while the user may be wanting to use a public  
> GRUU, they may not wish to announce their IMEI. Normally, no matter  
> what privacy, IMEIs are kept secret in 3GPP.
>

So it's rather like the distinction between appearing in public, and  
appearing naked in public ? One is acceptable for most, the other only  
for certain celebrities, or on just the right kind of beach . . .

--
Dean



From dean.willis@softarmor.com  Mon Feb 23 08:16:58 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Usable version of xml2rfc posted
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The debate around RFC 5378 has resulted in a delay on getting a  
version of xml2rfc out that will produce text suitable for submission.  
Cullen has apparently hacked a workaround, and we expect a  
"production" version in a few days.

Please get those drafts revised and submitted.

Cullen says:


> The XML2RFC folks are busy producing a good version of xml2rfc that  
> deals with the new license statements. However in the meantime I  
> have a hacked up version that does seem to work at least some of the  
> time.
>
> You can find it at
> http://svn.resiprocate.org/rep/ietf-drafts/xml2rfc/1.34pre2-fluffy/xml2rfc.tcl
>
> Hopefully this will soon be obsoleted when the real xml2rfc tool  
> gets updated.
>
> If you are trying to do a draft that has pre 5378 text in it, you  
> can set the ipr attribute in the <rfc> element to  
> ipr="pre5378Trust200902" and it will produce a txt file that id  
> submission tool will accept.
>
> Example xml at http://www.ietf.org/internet-drafts/draft-jennings-http-srv-01.xml 
>  and resulting txt file at
> http://www.ietf.org/internet-drafts/draft-jennings-http-srv-01.txt
>



From dean.willis@softarmor.com  Mon Feb 23 08:32:58 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] SSLstrip: Implications for SIPS and our imaginary padlock icon?
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I've often railed about how using "sips:" doesn't really prove  
anything to the user, since the connection is not end-to-end and  
downstream elements might be lying about what they are doing.

Apparently a similar attack is possible on https, when augmented with  
a (very) little social engineering. I'm pretty sure this trick would  
fool me occasionally, and my mother wouldn't stand a chance.



See:

http://www.forbes.com/2009/02/18/black-hat-hackers-technology-security_0218_blackhat.html



A poignant quote:

> The fundamental lesson of his encryption-stripping attack, says  
> Marlinspike, is that the protections on the Web's "secure" pages are  
> really just as weak as any page that can impersonate that security.  
> "The real answer is to encrypt everything," Marlinspike says. "When  
> you have a secure protocol that depends on an insecure protocol,  
> that's a problem."


So, what are the implications for the SIP world?

--
Dean


From theo@crazygreek.co.uk  Mon Feb 23 09:44:52 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
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Subject: Re: [Sip] SSLstrip: Implications for SIPS and our imaginary padlock icon?
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On Mon, Feb 23, 2009 at 4:33 PM, Dean Willis <dean.willis@softarmor.com> wrote:

> So, what are the implications for the SIP world?

Well, for starters vendors need to make sure any UI that represents
"this call is secure" display can't be subject to the favicon.ico
trick some browsers have.  think:

  Call-Info: <http://evil.attacker.com/padlock.jpg>;purpose=icon

SIP UIs also need to ensure that they can't fall for other
display-name tricks, i.e:

  From: "HSBC Bank"
<sips:mybankmanager%40hsbc.co.uk%3E%20%20%20%20%20%20%20%20%20%20%20%20@attacker.com>;tag=xxx

(it would appear the phone on my desk will indeed fall for something like this)

The SSLstrip attack relies on a MITM being able to modify a HTTP
response *before* getting to the HTTPS page (i.e, rewriting the links
a user would normally click, such as "<a
href="https://login.hsbc.co.uk">My Account</a>").

A parallel to this attack in SIP is if a SIPS URI is first learnt
through a non TLS requests, which upgrades to TLS and uses a UI hack
like the one above, or even redirect to a sips URI hosted on the
attackers server (resulting in a padlock, all be it the wrong one),
which then forwarded the call to le bank.  again, this is partially a
UI design issue, and we need to ensure that UAC vendors display who
the certificate has been issued to [1] as well at the identty of the
remote endpoint.

Luckily for us, the same sort of attack will be far harder in SIP once
DNSSEC is in place (if it ever is :)) in the end to end model, as we
use NAPTR for indicating the transport, this there is no risk of
rewriting the response, as it will come through TLS even when non sips
call in a "secured" domain;  HTTPS doesn't have this.  a User typing
in "mymanager@hsbc.co.uk" will - at least in the end to end model -
result in a NAPTR RRset for hsbc.co.uk's SRV records and only TLS
transports (assuming a bank was sensible and only has TLS transports),
which results in another (secured by DNSSEC) SRV lookup lookup for
their proxies, which then is sent via TLS to them.

The worry comes when a user types "sip:mymanager@hsbc.co.uk" which
goes over TCP/UDP (non TLS) to a service provider, which then could
return a 302 to sips:mymanager@hsbc.co.uk, as then we fall into the
same trap as the SSLstrip MITM attack.

In other words, a SIP call initiated over a non TLS transport can
never be "upgraded" (via, for example, a 3xx) to TLS without being
subject to this (albeit lack of user-awareness/UI design) attack.

 ~ Theo

1 - Although this has some issues too, obviously.

-- 
Theo Zourzouvillys
Chief Technical Officer
VoIP.co.uk

Sent from: Bicester Oxfordshire United Kingdom.

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On Mon, 2009-02-23 at 10:49 +0000, Theo Zourzouvillys wrote:
> On Sun, Feb 22, 2009 at 11:55 PM, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:
> 
> > Yup, these are essentially the same as IP-PBX trunks.  I don't know
> of many that allow requests in from truly unknown sources, though -
> they usually know the remote devices in advance, even when ENUM is
> used - so if they wanted to they could use IPSec or TLS if this became
> a problem. (some do today, but many don't - it seems many peer over
> private connections anyway)
> 
> I believe (hope!) that these are going to get more common!

Yes, Hadriel, you're thinking in a rather PSTN way.  Suppose I change a
few words in your sentence:  "I don't know of many Web servers that
allow requests in from truly unknown sources."  In the Pingtel days, our
gateway SBC would accept SIP calls from anywhere across the great
Internet.  Our current IT administration is a bit more backward.  But
we'll convince them eventually!

Dale



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On Fri, 2009-02-20 at 23:34 +0000, Theo Zourzouvillys wrote:
> > More importantly, such a value-less parameter does nothing
> > useful:  If the client will accept requests without cookies, a request
> > without the parameter will be accepted.  If the client will not accept
> > requests without cookies, a request without the parameter will elicit a
> > 4XX response, which can be used to resend the request.
> 
> A server could behave differently when it receives a request without a
> cookie in it, for example require authentication, redirect the user
> statelessly to another transport, or something we've not thought of
> just yet :-)

"A server may choose to apply additional security mechanisms to a
request without a cookie parameter."

> It's also useful for diagnostic purposes for deployment: how will we
> know when adoption is so widespread it would be safe to require
> support for it for a request to proceed?

Good point.

It would help if you added these points to the draft.

Dale



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Subject: [Sip] Diagnostic idea
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I've been considering the following idea for a diagnostic tool, and I'd
like to get some feedback on it.

The goal is to be able to trace the progress of a SIP request through
the network, including seeing the forking structure.  We first need to
pick a provisional response codes.  It appears that "170" is not
currently used.  This response code is also used as an option-tag for
this feature.  The processing is that whenever a SIP element receives a
request that contains "Supported: 170", the element will immediately (in
addition to anything else that it would do with the request) send a 170
response upstream, containing the request as its body (media type
message/sipfrag).

Dale



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From: "Elwell, John" <john.elwell@siemens.com>
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Dale,

History-Info can provide some information, depending of course whether
all proxies forward it and add to it as it goes downstream, that the UAS
sends it back, and that all proxies forward it upstream in the response.
It doesn't give you all the information you would get from your idea, of
course. Both mechanisms would suffer from border elements that strip
things out for topology hiding or other reasons.

On the detail, I believe 170 would establish an early dialog - is this
desirable, and if not is it avoidable?=20

John=20

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On=20
> Behalf Of Dale Worley
> Sent: 23 February 2009 21:37
> To: SIP
> Subject: [Sip] Diagnostic idea
>=20
> I've been considering the following idea for a diagnostic=20
> tool, and I'd
> like to get some feedback on it.
>=20
> The goal is to be able to trace the progress of a SIP request through
> the network, including seeing the forking structure.  We first need to
> pick a provisional response codes.  It appears that "170" is not
> currently used.  This response code is also used as an option-tag for
> this feature.  The processing is that whenever a SIP element=20
> receives a
> request that contains "Supported: 170", the element will=20
> immediately (in
> addition to anything else that it would do with the request)=20
> send a 170
> response upstream, containing the request as its body (media type
> message/sipfrag).
>=20
> Dale
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www.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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To: Dale Worley <dworley@nortel.com>, Theo Zourzouvillys <theo@voip.co.uk>
Date: Mon, 23 Feb 2009 18:34:26 -0500
Thread-Topic: [Sip] SIP amplification vulnerability -01 (draft-zourzouvillys-sip-via-cookie-01)
Thread-Index: AcmV/AjH9ujHk51gSdWF4sLZjJ3ZEwADIO0Q
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Cc: SIP WG <sip@ietf.org>
Subject: Re: [Sip] SIP amplification vulnerability -01	(draft-zourzouvillys-sip-via-cookie-01)
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> -----Original Message-----
> From: Dale Worley [mailto:dworley@nortel.com]
> Sent: Monday, February 23, 2009 4:18 PM
>
> On Mon, 2009-02-23 at 10:49 +0000, Theo Zourzouvillys wrote:
> > On Sun, Feb 22, 2009 at 11:55 PM, Hadriel Kaplan
> <HKaplan@acmepacket.com> wrote:
> >
> > > Yup, these are essentially the same as IP-PBX trunks.  I don't know
> > of many that allow requests in from truly unknown sources, though -
> > they usually know the remote devices in advance, even when ENUM is
> > used - so if they wanted to they could use IPSec or TLS if this became
> > a problem. (some do today, but many don't - it seems many peer over
> > private connections anyway)
> >
> > I believe (hope!) that these are going to get more common!
>
> Yes, Hadriel, you're thinking in a rather PSTN way.

No I'm not.  I'm merely indicating what I believe the common scenarios are =
as I see them today.  It has no relation to how my neurons interact.

-hadriel

From lakshmi@multitech.co.in  Mon Feb 23 20:52:23 2009
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From: lakshmi <lakshmi@multitech.co.in>
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Subject: [Sip] Handling MESSAGE with in dialog
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Hi,

We are enhancing SIP stack with MESSAGE method.  According to the 
MESSAGE method spec 3428, handling MESSAGE receive/send outside the 
dialog is clear. But 3428 is not clear in handling MESSAGE within the 
session. We are not clear about  the constraints like on which states 
MESSAGE events like receive/send MESSAGE must be allowed. Kindly clarify 
how to handle MESSAGE with in session/dialog.

 Regards,
-Lakshmi


From drage@alcatel-lucent.com  Tue Feb 24 02:56:28 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: "sip@ietf.org" <sip@ietf.org>
Date: Tue, 24 Feb 2009 11:56:38 +0100
Thread-Topic: Notes of identity conference call held 17th Feb 2009
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Subject: [Sip] Notes of identity conference call held 17th Feb 2009
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Attached find the notes of the last conference call of the interested parti=
es on progressing the identity issues in SIP, held on 17th Feb 2009. Expect=
 a revised version of draft-ietf-sip-e2e-identity-important shortly.

If anyone is interested in joining these discussions, send me a mail and I =
will include you in the details for the next call - we currently expect one=
 more before the IETF meeting.

regards

Keith

---------------------------------------------------------------------------=
---
Present:
Keith Drage
Hannes Tschofenig
Dan Wing
John Elwell
Dan York

>From the previous meeting notes:

> Ihe call came up with some areas where the document should be revised=20
> and John would attempt to capture those suggestions in a revised=20
> version of the document. Target date for new version of the document=20
> was 10th February so it could be discussed on our next call on 17th=20
> February. John made a plea for input to the draft in the form of words=20
> targetted at the draft.

So far John had not issued the revised version (draft-ietf-sip-e2e-identity=
-important) as he had not yet had sufficient input for the revised version.=
 A reminder on the mail list shortly before the call had produced some idea=
s.

John presented the changes he had made so far to the draft, which had not y=
et been circulated.

John had added a table at the end of the use cases part of the document as =
requested in the last call. The conclusion had also been split into a concl=
usion for email style identities and one for E164 style identities, also as=
 requested in the last call. A statement would also be added that end-to-en=
d identity is not achievable where the PSTN is involved.

John commented that he had received comments from David Schartz relating to=
 better trying to assign permitted operations on URIs to definitions of "co=
nvert", "translate" and "normalise". "translate" was understood to include =
adding a country code or conversion to alias. "normalise" was understood to=
 include e.g. addition or deletion of whitespace.

There was a requirement from the previous call to go into a greater detail =
of what could or could not be changed (posted 17 Feb 2009). This level of d=
etail was thought to be what Jon Peterson had been requesting. The remainde=
r of the call was spent reviewing the list that John had already posted to =
the participants. Hadrial had responded to this mail and his comments from =
this mail and prior mails were included in that review. It was noted that H=
adriel had not yet conducted a full review of this text, merely suggested t=
hat some previous mails also addressed these issues.

It was agreed that the focus was on fields signed by RFC 4474.

This combined text is reproduced below (with interpolated discussion from t=
he conference call):

> From last week's call, there was an action on all of us:
> "There was agreement that we should analyse in more detail specific=20
> header fields and body types, and the impact of SBCs on the identity=20
> characteristics of those elements. This will need to go into detail on=20
> the contents, for example, rather than a blanket change of a URI to a=20
> different URI, what is the impact of changing a URI to say an alias of=20
> that URI, or adding a "+" where one currently does not exist. Hadriel=20
> generally called these changes "protocol repair"."
>=20
> In the absence of any other contributions, I will try to kick this=20
> off. I don't known whether this was the sort of thing expected.
>=20
> Only those fields signed by RFC 4474 are considered.
>=20
> addr-spec of From header field:
> - Changing the host part means that the recipient loses information=20
> about the domain the request originated in.
> - Changing a non-E.164-based user-info part, or changing an=20
> E.164-based user-info part in ways not covered below, means that the=20
> recipient loses information about the particular user within the=20
> domain.
> - Changing digits within an E.164-based user part so to give a=20
> different generally accepted form of an E.164 number does not in=20
> principle remove the identification properties of From, although it=20
> could cause difficulties with URI matching at the destination. In=20
> generally, changing to a fully qualified, canonical form is less=20
> problematic than other changes (e.g., adding a "+" to an international=20
> number, removing dial string digits such as "00", adding country code.
> - Adding or removing user=3Dphone. In theory shouldn't be done, but in=20
> practice it normally does not remove the identification properties of=20
> From, since a the same user-part with/without user=3Dphone generally=20
> does identify different users.

There is a need to cover the cases of changes from E164 based to email styl=
e addresses and vice versa.

It was suggested that we could benefit here from David Schwartz's comments =
on categorising changes.=20

*** Does this constitute the same terminology from Jonathan's RAI area docu=
ment on numbering.

It was also suggested that we should avoid the document getting into a disc=
ussion of when is an E.164 number not an E.164 number.=20

> - Adding or removing other URI parameters. TODO.
>=20
SIP URI parameters have different characteristics to tel URI parameters in =
a number of ways and may need to be treated separately.

> addr-spec of To header field:
> - Not needed for identifying caller
> - For identifying original callee, considerations are similar to those=20
> for From
> - In particular, a change of To when a call is forwarded would destroy=20
> the identification properties of To.
>=20
Changing the To header field will break the RFC 4474 signature. It was poin=
ted out that independent of this, we should not be changing the To header f=
ield anyway, as doing so breaks the purpose of the To header field as defin=
ed in RFC 3261.

> call-id of Call-Id header field:
> - This does not impact the identity characteristics of a request.
>=20
> digit of CSeq header field:
> - This does not impact the identity characteristics of a request.
[JRE] Although Call-Id and CSeq were apparently included in the signature f=
or replay protection purposes, it is not clear that this would work. There =
are cases related to forking, redirection and retransmission over UDP trans=
port where the verifier could legitimately receive repeated requests with t=
he same Call-Id and CSeq values, perhaps with different values in the Reque=
st-URI.

>=20
> method of CSeq header field:
> - This does not impact the identity characteristics of a request. It=20
> seems very unlikely that an SBC would modify this field.
>=20
> Date header field:
> - This does not impact the identity characteristics of a request,=20
> although it is essential to provisions for preventing replay attacks.
>=20
> addr-spec of Contact header field:
> - Substitution of the SBC's own contact URI does not impact the=20
> identity characteristics of a request.
[JRE] A man-in-the-middle can pervert the routing of subsequent mid-dialog =
requests by manipulating Record-Route rather than Contact, so signing the C=
ontact header field has limited value in this respect. It may have some val=
ue for subsequent out-of-dialog requests (e.g., where the Contact URI is us=
ed in the Request-URI of a REFER request. In such cases, any B2BUA that mod=
ifies the Contact URI would need an algorithmic or stateful means of perfor=
ming reverse modification even after the original dialog has terminated. It=
 is not clear whether this works in practice.
Note that one reason for changing the contact URI is to replace IP addresse=
s, the use of which is bad practice anyway.

> - Changes to the original contact URI without substitution of the=20
> SBC's own contact URI is likely to be harmful for other reasons.
>=20
Some discussion is required of Contact header field parameters, e.g. the GR=
UU Contact header field parameter is the real identity for getting back to =
the user.

> Body parts other than SDP:
> - This does not impact the identity characteristics of a request,=20
> although it certainly has other implications.
> Unclear whether this happens in practice.
>=20
> SDP:
> - Reformatting (e.g., addition or removal of whitespace).=20
> This does not impact the identity characteristics of a request.
> - IP address in c-line / port in m-line. This impacts the identity=20
> characteristics of a request to the extent that it no longer=20
> identifies binds the sink of media to the user identified in From.
> However, SRTP with an appropriate key management system should achieve=20
> that and a lot more. The RTP case is insecure anyway.
> - a=3Dfingerprint line. This prevents the binding of SRTP to signalling=20
> and the user identified by the From URI.
> - removal of m-lines and their attributes. This does not impact the=20
> identity characteristics of a request

It os imderstood that the RTP case is insecure anyway. It was commented tha=
t Jon Peterson does not want the identity to be dependent on SRTP. Dan Wing=
 commented that even if SRTP is not secure, then identity is still useful e=
ven with RTP. We are frequently interested in who is calling and not necess=
sarily worried that some third party may have inserted an advert in the med=
ia stream.

> - Changing m-line transport, e.g., from SAVP to AVP. This does not=20
> impact the identity characteristics of a request, but constitutes a=20
> bid-down attack.
> - Removal of a=3Dtcap, a=3Dpcfg lines. Again this could constitute a=20
> bid-down attack.
>=20
> There are lots more things in SDP that could be changed. I won't go=20
> into that now. I first need some feedback on whether this is going in=20
> the right direction, and also advice on what in practice is likely to=20
> be changed for good reason at SBCs.
>=20
The document should cover the change of IP address.=20

There was also discussion of the of the change from AVPF to AVP, and the po=
tential impact of the use of capability negotiation to enable this. Suggest=
ed that feedback should be at the same level as the AVP.

Changing SDP is not necessarily bad or harmful. It was understood that the =
area directors wanted to sign the SDP because IETF does not necesssarily un=
derstand what might be used in the future in the SDP. Therefore we sign the=
 SDP except all those things we allow to change.

> Although many of the fields can be changed without impacting the=20
> identity characteristics of a request, the reason they are signed in=20
> RFC 4474 is to do with providing a measure of integrity protection on=20
> important parts of the message that are not expected to change. The=20
> more of these we remove from the signature, the less integrity=20
> protection we have (and in some case less playback protection). So we=20
> need some sort of balance between allowing reasonable changes by SBCs=20
> while still giving a reasonable measure of integrity and playback=20
> protection. We also have to take into account that some requests do=20
> not relate to media and do not contain SDP (e.g., MESSAGE, SUBSCRIBE,=20
> NOTIFY), and therefore reducing the amount of signed information will=20
> impact the integrity and playback protection of these requests more=20
> than requests such as INVITE.

There was a discussion of other information that also should be discussed. =
The following were suggested:

-	Geolocation and any associated PIDF LO body.
-	SAML assertion.

Both would be signed by the XML signature.

The document being produced needs to make sure we clearly identify what we =
discussed and excluded from the discussion. There is a need to justify what=
 is left is the complete list that merits discussion, and that nothing has =
therefore been missed.

This was the last of the currently scheduled calls. It was agreed to have a=
 further call on Tuesday 3rd March 2009 at 7:00 a.m. Pacific time.

---------------------------------------------------------------------------=
---

regards

Keith=

From ekr@networkresonance.com  Tue Feb 24 08:00:27 2009
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Date: Tue, 24 Feb 2009 08:23:30 -0800
From: Eric Rescorla <ekr@networkresonance.com>
To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] SSLstrip: Implications for SIPS and our imaginary padlock	icon?
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At Mon, 23 Feb 2009 10:33:07 -0600,
Dean Willis wrote:
> 
> 
> I've often railed about how using "sips:" doesn't really prove  
> anything to the user, since the connection is not end-to-end and  
> downstream elements might be lying about what they are doing.
> 
> Apparently a similar attack is possible on https, when augmented with  
> a (very) little social engineering. I'm pretty sure this trick would  
> fool me occasionally, and my mother wouldn't stand a chance.
> 
> 
> 
> See:
> 
> http://www.forbes.com/2009/02/18/black-hat-hackers-technology-security_0218_blackhat.html
> 
> 
> 
> A poignant quote:
> 
> > The fundamental lesson of his encryption-stripping attack, says  
> > Marlinspike, is that the protections on the Web's "secure" pages are  
> > really just as weak as any page that can impersonate that security.  
> > "The real answer is to encrypt everything," Marlinspike says. "When  
> > you have a secure protocol that depends on an insecure protocol,  
> > that's a problem."
> 
> 
> So, what are the implications for the SIP world?

Uh, none? 

This is the whole reason why you need to start with secure references
and not allow opportunistic upgrade.

-Ekr

From rjsparks@nostrum.com  Tue Feb 24 08:02:46 2009
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Won't this run into exactly the same response-size obstacle that the  
previous diagnostics work tripped up on?
It might help to spend some time trying to find some other pattern  
than trying to use responses for moving the information you're trying  
to capture around.
Source-routed requests in the opposite direction? (using content  
indirection?) Subscribe/Notify? Maybe something else...

RjS

On Feb 23, 2009, at 3:37 PM, Dale Worley wrote:

> I've been considering the following idea for a diagnostic tool, and  
> I'd
> like to get some feedback on it.
>
> The goal is to be able to trace the progress of a SIP request through
> the network, including seeing the forking structure.  We first need to
> pick a provisional response codes.  It appears that "170" is not
> currently used.  This response code is also used as an option-tag for
> this feature.  The processing is that whenever a SIP element  
> receives a
> request that contains "Supported: 170", the element will immediately  
> (in
> addition to anything else that it would do with the request) send a  
> 170
> response upstream, containing the request as its body (media type
> message/sipfrag).
>
> Dale
>
>
> _______________________________________________
> Sip mailing list  https://www.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 dean.willis@softarmor.com  Tue Feb 24 08:33:38 2009
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On Feb 23, 2009, at 3:37 PM, Dale Worley wrote:

> I've been considering the following idea for a diagnostic tool, and  
> I'd
> like to get some feedback on it.
>
> The goal is to be able to trace the progress of a SIP request through
> the network, including seeing the forking structure.  We first need to
> pick a provisional response codes.  It appears that "170" is not
> currently used.  This response code is also used as an option-tag for
> this feature.  The processing is that whenever a SIP element  
> receives a
> request that contains "Supported: 170", the element will immediately  
> (in
> addition to anything else that it would do with the request) send a  
> 170
> response upstream, containing the request as its body (media type
> message/sipfrag).


Your idea has merit. I believe Jiri Kuthan was working on something  
like this a few years ago.

I've also previously suggested a similar provisional response to  
inform the UAC about retargeting being done by a proxy. In addition to  
debugging, this helps avoid the unanticipated respondent problem.

--
Dean

From dean.willis@softarmor.com  Tue Feb 24 08:36:36 2009
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On Feb 24, 2009, at 10:03 AM, Robert Sparks wrote:

> Won't this run into exactly the same response-size obstacle that the  
> previous diagnostics work tripped up on?
> It might help to spend some time trying to find some other pattern  
> than trying to use responses for moving the information you're  
> trying to capture around.
> Source-routed requests in the opposite direction? (using content  
> indirection?) Subscribe/Notify? Maybe something else...


True.

Do we really need the whole message, or just enough information to  
identify the message and its context, and possibly describe what the  
proxy is doing in response to the message?

For example, what about returning a response that contains little more  
than the minimum plus a copy of the H-I header that would have been  
added?

--
Dean


From ekr@networkresonance.com  Tue Feb 24 08:58:28 2009
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Subject: Re: [Sip] Notes of identity conference call held 17th Feb 2009
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At Tue, 24 Feb 2009 11:56:38 +0100,
DRAGE, Keith (Keith) wrote:
> 
> Attached find the notes of the last conference call of the
> interested parties on progressing the identity issues in SIP, held
> on 17th Feb 2009. Expect a revised version of
> draft-ietf-sip-e2e-identity-important shortly.

draft-IETF...?

Did this get made a WG document when I wasn't paying attention?

-Ekr

From adam@nostrum.com  Tue Feb 24 09:12:03 2009
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Theo:

Thanks for pulling this draft together. I certainly admire the problem 
you're trying to solve, and you've done a very good job of summarizing 
it and its effects.

I have one fundamental point about the proposed mechanism, and a 
suggestion for further analysis.

The fundamental point is based on your observation that this attack can 
be effective when the victim is a non-SIP endpoint, simply because the 
incoming SIP messages can saturate their link. (I'll add to that that 
the attack can wreak havoc on a deployed non-SIP service by 
intentionally targeting that service's port). With this fact in mind, 
It's not clear to me how the proposed extension overcomes the described 
problem. To wit:

   1. The attacker sends a request to the amplifiers, with a spoofed
      source IP.

   2. Each of the amplifiers, which are compliant to this specification,
      sends the new 4XX defined by this document to the victim's IP
      address. It contains a cookie.

   3. Of course, the victim (not having a SIP client running on that
      port) doesn't respond (at least, not with a SIP message). The
      attacker, wanting his attack to succeed, has craftily chosen a
      port he knows to be open (probably by first portscanning the
      victim). As a consequence, the victim does not send any ICMP error
      to the amplifiers.

   4. Because they didn't get an ACK, 500 ms later, each of the
      amplifiers sends another 4XX response.

   5. This continues according to the response retransmission rules of
      RFC 3261 until all 11 responses have been sent.

So the 11-to-1 amplification attack doesn't appear to be fixed by this 
mechanism, assuming the attacker makes a rather modest attempt to succeed.

Presuming this can be remedied (and I have no immediate suggestions for 
doing so), I would recommend you consider the interaction between your 
proposal and forking. Analyze it in the context of the HERFP issues, and 
make sure forking doesn't cause unintended behavior when interacting 
with your mechanism. Also, consider longer proxy chains. For example:

   Note that a 4XX response MUST trigger an ACK to be sent as per normal
   SIP processing rules.  The generated ACK MUST contain the cookie
   parameter copied from the 4XX response.

Keeping in mind that ACK messages triggered by non-2XX responses are 
hop-by-hop, this falls down when you have a proxy between the UAC and 
the server sending the 4XX. The intervening proxy may not implement this 
extension, so you can't rely on it doing anything special with the ACK 
(unless you define a Proxy-Require, which pretty much renders the 
extension undeployable).

/a

From drage@alcatel-lucent.com  Tue Feb 24 09:40:38 2009
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No, just me suffering from advanced jet lag having flown back into the UK o=
ver the weekend.

It is of course draft-elwell-sip-e2e-identity-important that you will be ex=
pecting a revised version of.=20

regards

Keith

> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@networkresonance.com]=20
> Sent: Tuesday, February 24, 2009 5:22 PM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Notes of identity conference call held=20
> 17th Feb 2009
>=20
> At Tue, 24 Feb 2009 11:56:38 +0100,
> DRAGE, Keith (Keith) wrote:
> >=20
> > Attached find the notes of the last conference call of the=20
> interested=20
> > parties on progressing the identity issues in SIP, held on 17th Feb=20
> > 2009. Expect a revised version of=20
> > draft-ietf-sip-e2e-identity-important shortly.
>=20
> draft-IETF...?
>=20
> Did this get made a WG document when I wasn't paying attention?
>=20
> -Ekr
> =

From theo@crazygreek.co.uk  Tue Feb 24 10:20:17 2009
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From: Theo Zourzouvillys <theo@crazygreek.co.uk>
To: Adam Roach <adam@nostrum.com>
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Cc: SIP WG <sip@ietf.org>, draft-zourzouvillys-sip-via-cookie@tools.ietf.org
Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie
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On Tue, Feb 24, 2009 at 5:12 PM, Adam Roach <adam@nostrum.com> wrote:

Hi Adam,

Thank you for the feedback;

> With this fact in mind, It's not clear to me how the
> proposed extension overcomes the described problem. To wit:
>
> *snip*
>
> =A05. This continues according to the response retransmission rules of
> =A0 =A0 RFC 3261 until all 11 responses have been sent.
>
> So the 11-to-1 amplification attack doesn't appear to be fixed by this
> mechanism, assuming the attacker makes a rather modest attempt to succeed=
.

The key here is that the 4XX response to a request without a cookie is
sent by the next hop statelessly, and doesn't cause a server
transaction to be created (and thus no re-transmits); as per 4.3, para
4:

   If a request is received from an IP address that could be spoofable
   (i.e, any request received from the general internet), and that
   request is going to have a server transaction created for it, and the
   top via field contains a cookie parameter, then the server SHOULD
   generate a new cookie for this source and generates a 4XX response,
   and place the cookie value into the cookie parameter of the top via
   field before sending the response statelessly.

I'll update this in the working draft to also include:

 ... and MUST NOT create a server transaction for the request.  This implie=
s
  that a server implementing this draft MUST NOT send a 100 response before
  sending the 4XX.

does this satisfy your concern?

> Presuming this can be remedied (and I have no immediate suggestions for
> doing so), I would recommend you consider the interaction between your
> proposal and forking. Analyze it in the context of the HERFP issues, and
> make sure forking doesn't cause unintended behavior when interacting with
> your mechanism.

Because of the hop-by-hop semantics, i can't see any issues that could
be caused with the cookie as any 4XX response is just re-transmitted
with the cookie, or if the client doesn't support it and the server
requires it then it will just be treated like a 400 response -
although i'll have more of a think about any issues it could cause in
case there are any unintended side effects.

> Also, consider longer proxy chains. For example:
>
> =A0Note that a 4XX response MUST trigger an ACK to be sent as per normal
> =A0SIP processing rules. =A0The generated ACK MUST contain the cookie
> =A0parameter copied from the 4XX response.
>
> Keeping in mind that ACK messages triggered by non-2XX responses are
> hop-by-hop, this falls down when you have a proxy between the UAC and the
> server sending the 4XX. The intervening proxy may not implement this
> extension, so you can't rely on it doing anything special with the ACK
> (unless you define a Proxy-Require, which pretty much renders the extensi=
on
> undeployable).

In the case a previous hop doesn't support this draft, then the server
sending the 4XX would know that the client didn't support the cookie
anyway, because the request's top via header didn't contain an empty
"cookie" parameter. Although the copying-to-the-ACK is not there for
any reason other than diagnostic purposes right now - although
arguably the server sending the 4XX can also set the To tag to contain
any data it wanted to store so could be removed.

 ~ Theo

From scott.lawrence@nortel.com  Tue Feb 24 10:30:48 2009
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Subject: Re: [Sip] Implementing SHA digest auth (new topic)
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On Tue, 2009-01-06 at 15:33 +0100, Nils Ohlmeier wrote:
> >
> But I think if we take that road we should consider some mechanism which
> allows UAC and UAS to advertise and agree (?) on the most preferred
> algorithm for the authentication. And I guess it should be mandatory that
> this mechanism prevents that an attacker can modify the message exchange
> in a way that the weakest algorithm is chosen (which is the case with
> current proposed solution).

The mechanism exists.  When it sends the challenge, the server sends a
qop parameter whose value is the list of methods it supports, and when
the client sends authentication, the qop parameter indicates which
method it used.

All that is needed is to define new qop token values and how they map to
a new hash algorithm.



From adam@nostrum.com  Tue Feb 24 13:29:36 2009
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Cc: SIP WG <sip@ietf.org>, draft-zourzouvillys-sip-via-cookie@tools.ietf.org
Subject: Re: [Sip] draft-zourzouvillys-sip-via-cookie
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Theo Zourzouvillys wrote:
> On Tue, Feb 24, 2009 at 5:12 PM, Adam Roach <adam@nostrum.com> wrote:
>
> Hi Adam,
>
> Thank you for the feedback;
>
>   
>> With this fact in mind, It's not clear to me how the
>> proposed extension overcomes the described problem. To wit:
>>
>> *snip*
>>
>>  5. This continues according to the response retransmission rules of
>>     RFC 3261 until all 11 responses have been sent.
>>
>> So the 11-to-1 amplification attack doesn't appear to be fixed by this
>> mechanism, assuming the attacker makes a rather modest attempt to succeed.
>>     
>
> The key here is that the 4XX response to a request without a cookie is
> sent by the next hop statelessly, and doesn't cause a server
> transaction to be created (and thus no re-transmits); as per 4.3, para
> 4:
>
>    If a request is received from an IP address that could be spoofable
>    (i.e, any request received from the general internet), and that
>    request is going to have a server transaction created for it, and the
>    top via field contains a cookie parameter, then the server SHOULD
>    generate a new cookie for this source and generates a 4XX response,
>    and place the cookie value into the cookie parameter of the top via
>    field before sending the response statelessly.
>   

Ah -- "statelessly" means something very different to you than it does 
to me. I think you need to explicitly call out that you're proposing a 
modification to the basic INVITE transaction model defined in RFC 3261 
for this new response.

This, combined with prohibiting provisional responses, would seem to 
have the desired effect.

>> Also, consider longer proxy chains. For example:
>>
>>  Note that a 4XX response MUST trigger an ACK to be sent as per normal
>>  SIP processing rules.  The generated ACK MUST contain the cookie
>>  parameter copied from the 4XX response.
>>
>> Keeping in mind that ACK messages triggered by non-2XX responses are
>> hop-by-hop, this falls down when you have a proxy between the UAC and the
>> server sending the 4XX. The intervening proxy may not implement this
>> extension, so you can't rely on it doing anything special with the ACK
>> (unless you define a Proxy-Require, which pretty much renders the extension
>> undeployable).
>>     
>
> In the case a previous hop doesn't support this draft, then the server
> sending the 4XX would know that the client didn't support the cookie
> anyway, because the request's top via header didn't contain an empty
> "cookie" parameter. Although the copying-to-the-ACK is not there for
> any reason other than diagnostic purposes right now - although
> arguably the server sending the 4XX can also set the To tag to contain
> any data it wanted to store so could be removed.
>   

It sounds like you're currently setting this up so that it can't work 
transitively -- that is, if you have a chain of proxies, some of them 
compliant to this behavior, others that aren't, then any proxy past a 
non-compliant one can't check return routability. That seems like an 
artificial limitation. If you had some other way of indicating support 
by a client that could be checked by every proxy in the chain, then you 
could perform routability checks all the way back to the client 
regardless of intervening older proxies.

/a

From dwing@cisco.com  Tue Feb 24 13:43:04 2009
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Cc: sip@ietf.org
Subject: Re: [Sip] Notes of identity conference call held 17th Feb 2009
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> > Attached find the notes of the last conference call of the
> > interested parties on progressing the identity issues in SIP, held
> > on 17th Feb 2009. Expect a revised version of
> > draft-ietf-sip-e2e-identity-important shortly.
> 
> draft-IETF...?
> 
> Did this get made a WG document when I wasn't paying attention?

Ekr,

Keith clarified that he made a mistake with the document name.

But your surprise that this could have become a WG document leads me to feel
that you believe end to end identity is not imporatnt.  Is my feeling correct,
and that you believe end to end identity is not important?

-d


From theo@crazygreek.co.uk  Tue Feb 24 13:58:04 2009
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On Tue, Feb 24, 2009 at 9:29 PM, Adam Roach <adam@nostrum.com> wrote:
> Ah -- "statelessly" means something very different to you than it does to
> me. I think you need to explicitly call out that you're proposing a
> modification to the basic INVITE transaction model defined in RFC 3261 for
> this new response.

will do

> It sounds like you're currently setting this up so that it can't work
> transitively -- that is, if you have a chain of proxies, some of them
> compliant to this behavior, others that aren't, then any proxy past a
> non-compliant one can't check return routability. That seems like an
> artificial limitation. If you had some other way of indicating support by a
> client that could be checked by every proxy in the chain, then you could
> perform routability checks all the way back to the client regardless of
> intervening older proxies.

While I did initially consider this problem, and indeed came up with a
potential solution in the form of a new authentication scheme or a
header that the client then used to insert in it's own via header,
HERFP as well as multi-hop authentication just seemed to make any
solution i could come up with very complex compared to this method -
which while it has it's downsides in terms of backward compatibility,
the beauty is in it's simplicity and that it should hopefully not be a
burden for stacks to add with no change outside of the
transaction/transport layer.

 ~ Theo

From shida@ntt-at.com  Tue Feb 24 21:59:21 2009
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Cc: "sip@ietf.org List" <sip@ietf.org>, draft-ietf-sip-ua-privacy@tools.ietf.org
Subject: Re: [Sip] Comment on sip-ua-privacy-05.txt
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Keith;

  So what consideration should be noted in the draft with regards to  
IMEI?

  I am reading that IMEI is never revealed, whether it's a public or  
temp GRUU,
that sounds like a non-privacy concern. And if it's not a privacy  
threat I don't
know what type of text be added..

  I am probably missing your point, but I think I need a bit of  
guidance here..

  Thanks
   Shida

On 24-Feb-09, at 12:43 AM, Dean Willis wrote:

>
> On Feb 23, 2009, at 1:30 AM, DRAGE, Keith (Keith) wrote:
>
>> The issue is that while the user may be wanting to use a public  
>> GRUU, they may not wish to announce their IMEI. Normally, no matter  
>> what privacy, IMEIs are kept secret in 3GPP.
>>
>
> So it's rather like the distinction between appearing in public, and  
> appearing naked in public ? One is acceptable for most, the other  
> only for certain celebrities, or on just the right kind of beach . . .
>
> --
> Dean
>
>


From michael@voip.co.uk  Tue Feb 24 23:27:32 2009
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Hi Adam,

Adam Roach wrote:
> Ah -- "statelessly" means something very different to you than it does
> to me. I think you need to explicitly call out that you're proposing a
> modification to the basic INVITE transaction model defined in RFC 3261
> for this new response.
>
> This, combined with prohibiting provisional responses, would seem to
> have the desired effect.

I don't think there is any proposed modification.  The response should
be sent according to the rules of RFC 3261 section 8.2.7.  Amongst other
things, this specifically prohibits provisional responses.

I think the draft should explicitly mention this section.

Regards,

Michael

From adam@nostrum.com  Wed Feb 25 07:00:58 2009
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Michael Procter wrote:
> Hi Adam,
>
> Adam Roach wrote:
>   
>> Ah -- "statelessly" means something very different to you than it does
>> to me. I think you need to explicitly call out that you're proposing a
>> modification to the basic INVITE transaction model defined in RFC 3261
>> for this new response.
>>
>> This, combined with prohibiting provisional responses, would seem to
>> have the desired effect.
>>     
>
> I don't think there is any proposed modification.  The response should
> be sent according to the rules of RFC 3261 section 8.2.7.

Wow -- I had completely forgotten about 8.2.7.

>   Amongst other
> things, this specifically prohibits provisional responses.
>
> I think the draft should explicitly mention this section.
>   


Yes, that would be a significant improvement. I think you'll also want 
to give the cookie the same properties as 8.2.7 describes for to-tags.

/a

From john.elwell@siemens.com  Wed Feb 25 12:21:00 2009
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Date: Wed, 25 Feb 2009 20:21:13 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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The posted draft contains a number of additions compared with 02, as
discussed within an informal group during the last few weeks:
- identifier modifications classified as conversion, translation or
normalisation (section 4);
- tabular summary of examples (section 5);
- analysis of changes to SIP requests by B2BUAs (section 8);
- extension to conclusions (section 9);
- a start at requirements (section 10).

http://www.ietf.org/internet-drafts/draft-elwell-sip-e2e-identity-import
ant-03.txt

John

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On Wed, 2009-02-25 at 07:27 +0000, Michael Procter wrote:
> Adam Roach wrote:
> > Ah -- "statelessly" means something very different to you than it does
> > to me. I think you need to explicitly call out that you're proposing a
> > modification to the basic INVITE transaction model defined in RFC 3261
> > for this new response.
> >
> > This, combined with prohibiting provisional responses, would seem to
> > have the desired effect.
> 
> I don't think there is any proposed modification.  The response should
> be sent according to the rules of RFC 3261 section 8.2.7.  Amongst other
> things, this specifically prohibits provisional responses.
> 
> I think the draft should explicitly mention this section.

And more importantly, according to 8.2.7, the recipient of an INVITE
does *not* resend a 4xx response if it doesn't receive an ACK.  At first
read, this seems to violate the 3-way-handshake model of the INVITE
transaction, but it doesn't -- If the 4xx got lost, the UAC will send it
again, prompting the UAS to send an identical 4xx response.  If the ACK
gets lost, the UAS doesn't care.

Dale



From dworley@nortel.com  Wed Feb 25 13:39:02 2009
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On Wed, 2009-02-25 at 09:01 -0600, Adam Roach wrote:
> Wow -- I had completely forgotten about 8.2.7.
> 
> > Amongst other
> > things, this specifically prohibits provisional responses.
> >
> > I think the draft should explicitly mention this section.
> >   
> 
> 
> Yes, that would be a significant improvement. I think you'll also want 
> to give the cookie the same properties as 8.2.7 describes for to-tags.

Hmmm...  Oddly, you can't do that, because the cookies should depend on
the time.  But the cookie mechanism quasi-satisfies 8.2.7, because
multiple sends of an INVITE will receive multiple responses with
different cookies, but all of the cookies are "equivalent", in that any
one of them can be chosen for successful future use.

Dale



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On Mon, 2009-02-23 at 22:08 +0000, Elwell, John wrote:
> History-Info can provide some information, depending of course whether
> all proxies forward it and add to it as it goes downstream, that the UAS
> sends it back, and that all proxies forward it upstream in the response.

Yes, if every element implements History-Info fully, you get even better
information than a flock of 170's.  But I originated this with the idea
of inserting it into the sipX stack, and it would take less work to
implement 170 than History-Info.  There's also the interesting idea of
collating the History-Info values from a bunch of 170's.

> Both mechanisms would suffer from border elements that strip
> things out for topology hiding or other reasons.

Yes.  And if I go forward with this idea, I would need to show that
topology hiding could be done easily and sensibly.  One case that would
be nice to handle well is that if the UAC's and UAS's organizational
networks did not want to topology-hide but the transit carrier did --
you'd like to be able to have the transit carrier hide its topology
without hiding what happens in the endpoint networks.

> On the detail, I believe 170 would establish an early dialog - is this
> desirable, and if not is it avoidable? 

It would establish an early dialog, and it's not avoidable, other than
for the fact that any element that understands 170's knows that there
will be no further messages in the early dialog.  I'm not sure if it
causes any problems.

I haven't thought through the interaction with 100rel, though.  I think
the way to handle that would be to postulate that the 170 came from an
internal UAS that did not implement 100rel.

On Tue, 2009-02-24 at 10:03 -0600, Robert Sparks wrote:
> Won't this run into exactly the same response-size obstacle that the  
> previous diagnostics work tripped up on?

I've mostly assumed that it would be used in an enterprise network, and
not worried about message size.  sipX has been implementing
draft-ietf-sip-hop-limit-diagnostics for years, and we haven't seen any
problems.  Although in the case of a provisional response, its loss
isn't much of a problem.

> It might help to spend some time trying to find some other pattern  
> than trying to use responses for moving the information you're trying  
> to capture around.
> Source-routed requests in the opposite direction? (using content  
> indirection?) Subscribe/Notify? Maybe something else...

Yes, that would help, though it's not clear what mechanism could be as
simple as a provisional response.

On Tue, 2009-02-24 at 10:36 -0600, Dean Willis wrote: 
> Do we really need the whole message, or just enough information to  
> identify the message and its context, and possibly describe what the  
> proxy is doing in response to the message?
> 
> For example, what about returning a response that contains little more  
> than the minimum plus a copy of the H-I header that would have been  
> added?

A good point.  Though it's hard to explicitly rule out any particular
header from being returned, given that in some particular case, the
proxies' transformations of that header is the interesting question.


On Tue, 2009-02-24 at 10:33 -0600, Dean Willis wrote: 
> Your idea has merit. I believe Jiri Kuthan was working on something  
> like this a few years ago.

http://tools.ietf.org/html/draft-kuthan-sipping-diag-00 -- Thanks, I'll
have to look at it.

> I've also previously suggested a similar provisional response to  
> inform the UAC about retargeting being done by a proxy. In addition to  
> debugging, this helps avoid the unanticipated respondent problem.

Which suggests the possibility of sending 170's based not on the receipt
of a request, but on *sending* a copy of the request downstream.  The
effect would not be much different, but it would require implementing
170 only in proxies.

It would also be helpful if the 170 carried the sending and receiving
address/port/transport triples, in addition to the usual request-URI.

Dale



From christer.holmberg@ericsson.com  Thu Feb 26 04:27:59 2009
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Hi,=20

>Glad to see that the draft starts its WGLC process.

Well, that is what I requested at least :)

>I still have some difficult in using IETF and cannot find the whole
sorted comments about this draft, so i have some doubt about this draft.
>=20
>   1.   "If a forking proxy receives a reliably sent 199 response for a
dialog, for which the proxy has previously generated and sent a 199
response, the proxy MUST forward the 199 response."
>
>     Does it describe the case below? Although P1 have sent a 199
response, P1 havs to  forword the send reliably 199 too.Or  is the
first 199 mistaked for 18x?
>=20
>          UAC               P1                  UAS_2   =20
>            --- INVITE ------>
>                             --- INVITE (leg 2) ->
>            <-- 199(leg 2) --
>                             <-- 199 (leg 2) -----
>            <-- 199(leg 2) --
=20
I think it shall be 199, as currently written.

>2.  "10.  Usage with 100rel=20
>
>   When a 199 Early Dialog Terminated provisional response is sent by a
UAS, since the provisional response is only used for information
purpose, the UAS SHOULD send it unreliably even if the 100rel=20
>   option tag [RFC3262] is present in the Require header of the
associated request."
>
>  =20
>I have seen a comment on this question,but still not understood about
it. If the INVITE has a Require tag "Require: 100rel",does the UAS still
use unreliable 199 response?

That is the recommendation, yes. The reasons is that we want to keep 199
as "lightweight" as possible, without requireing re-transmissions and
PRACKs.

>If 199 is reliable, there is one more advantage. If the 199 arrives
before the first 18x response, UAC can discard the first 199 and process
it until UAC receives the first 18x response that has the same=20
>to-tag as 199, as the reliable 199 should be re-transmited until
received PRACK.

If the INVITE contains "Require: 100rel", the first 18x must also be
sent reliably. And, I don't think the UAS is allowed to send another
reliable response until the first one is acknowledged, so I don't think
a reliable 199 would reach the UAC before the first reliable 18x.

>Can anyone show me how to sort all comments of a topic by time?

Unless it can be done by sorting the e-mails by subject it is kinda
difficult.

Regards,

Christer

=20

   =20

=20


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Hi Christer,
   Thanks for replying.
   Comments are maked with [Eric] below. 3ks.

>Glad to see that the draft starts its WGLC process.

Well, that is what I requested at least :)

>I still have some difficult in using IETF and cannot find the whole
sorted comments about this draft, so i have some doubt about this draft.
>
>   1.   "If a forking proxy receives a reliably sent 199 response for a
dialog, for which the proxy has previously generated and sent a 199
response, the proxy MUST forward the 199 response."
>
>     Does it describe the case below? Although P1 have sent a 199
response, P1 havs to  forword the send reliably 199 too.Or  is the
first 199 mistaked for 18x?
>
>          UAC               P1                  UAS_2
>            --- INVITE ------>
>                             --- INVITE (leg 2) ->
>            <-- 199(leg 2) --
>                             <-- 199 (leg 2) -----
>            <-- 199(leg 2) --

I think it shall be 199, as currently written.

[Eric]: If  this is 199, then Is there a special purpose to send two 199 in
the same dialog?
I think that one 199 is enough.
And  the first 199 may be reliable too, that will make it a little difficult
to send the second 199.


>2.  "10.  Usage with 100rel
>
>   When a 199 Early Dialog Terminated provisional response is sent by a
UAS, since the provisional response is only used for information
purpose, the UAS SHOULD send it unreliably even if the 100rel
>   option tag [RFC3262] is present in the Require header of the
associated request."
>
>
>I have seen a comment on this question,but still not understood about
it. If the INVITE has a Require tag "Require: 100rel",does the UAS still
use unreliable 199 response?

That is the recommendation, yes. The reasons is that we want to keep 199
as "lightweight" as possible, without requireing re-transmissions and
PRACKs.

[Eric]: But reliable 199 have more advantage, and It is worth to use the
reliable 199,I think.

>If 199 is reliable, there is one more advantage. If the 199 arrives
before the first 18x response, UAC can discard the first 199 and process
it until UAC receives the first 18x response that has the same
>to-tag as 199, as the reliable 199 should be re-transmited until
received PRACK.

If the INVITE contains "Require: 100rel", the first 18x must also be
sent reliably. And, I don't think the UAS is allowed to send another
reliable response until the first one is acknowledged, so I don't think
a reliable 199 would reach the UAC before the first reliable 18x.

[Eric]: Surely the UAS is allowed to send another reliable response until
the first one is acknowledged, but reliable 199 will be useful if the first
18x is unreliable, or
the first 18x has been acknwledged.
It's another case different from the above one  whose INVITE contains
"require: 100rel".


>Can anyone show me how to sort all comments of a topic by time?

Unless it can be done by sorting the e-mails by subject it is kinda
difficult.

Regards,

Christer


Regards,

Eric.wang

--000e0cd24b9ec22e7f0463d34a79
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Hi Christer,</div>
<div>=A0=A0 Thanks for replying.</div>
<div>=A0=A0 Comments are maked with [Eric] below. 3ks.<br><br>&gt;Glad to s=
ee that the draft starts its WGLC process.<br><br>Well, that is what I requ=
ested at least :)<br><br>&gt;I still have some difficult in using IETF and =
cannot find the whole<br>
sorted comments about this draft, so i have some doubt about this draft.<br=
>&gt;<br>&gt;=A0=A0 1.=A0=A0 &quot;If a forking proxy receives a reliably s=
ent 199 response for a<br>dialog, for which the proxy has previously genera=
ted and sent a 199<br>
response, the proxy MUST forward the 199 response.&quot;<br>&gt;<br>&gt;=A0=
=A0=A0=A0 Does it describe the case below? Although P1 have sent a 199<br>r=
esponse, P1 havs to=A0=A0forword the send reliably 199 too.Or=A0=A0is the<b=
r>first 199 mistaked for 18x?<br>
&gt;<br>&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0UAC=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 P1=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0UAS_2<=
br>&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0--- INVITE ------&gt;<br>&gt;=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 --- INVITE (leg 2) -&gt;<br>&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
&lt;-- 199(leg 2) --<br>&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 &lt;-- 199 (leg 2) -----<br>
&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0&lt;-- 199(leg 2) --<br><br>I think=
 it shall be 199, as currently written.<br>=A0</div>
<div><font color=3D"#ff0000">[Eric]:=A0If=A0 this is 199, then=A0Is there=
=A0a special purpose to send two 199 in the=A0same dialog?</font></div>
<div><font color=3D"#ff0000">I think that one 199 is enough.</font></div>
<div><font color=3D"#ff0000">And=A0 the first 199 may be reliable too, that=
 will make it a little difficult to send the second 199.</font></div>
<div>=A0</div>
<div><br>&gt;2.=A0=A0&quot;10.=A0=A0Usage with 100rel<br>&gt;<br>&gt;=A0=A0=
 When a 199 Early Dialog Terminated provisional response is sent by a<br>UA=
S, since the provisional response is only used for information<br>purpose, =
the UAS SHOULD send it unreliably even if the 100rel<br>
&gt;=A0=A0 option tag [RFC3262] is present in the Require header of the<br>=
associated request.&quot;<br>&gt;<br>&gt;<br>&gt;I have seen a comment on t=
his question,but still not understood about<br>it. If the INVITE has a Requ=
ire tag &quot;Require: 100rel&quot;,does the UAS still<br>
use unreliable 199 response?<br><br>That is the recommendation, yes. The re=
asons is that we want to keep 199<br>as &quot;lightweight&quot; as possible=
, without requireing re-transmissions and<br>PRACKs.</div>
<div>=A0</div>
<div><font color=3D"#ff0000">[Eric]: But reliable 199 have more advantage, =
and It is worth to use the reliable 199,I think.</font><br><br>&gt;If 199 i=
s reliable, there is one more advantage. If the 199 arrives<br>before the f=
irst 18x response, UAC can discard the first 199 and process<br>
it until UAC receives the first 18x response that has the same<br>&gt;to-ta=
g as 199, as the reliable 199 should be re-transmited until<br>received PRA=
CK.<br><br>If the INVITE contains &quot;Require: 100rel&quot;, the first 18=
x must also be<br>
sent reliably. And, I don&#39;t think the UAS is allowed to send another<br=
>reliable response until the first one is acknowledged, so I don&#39;t thin=
k<br>a reliable 199 would reach the UAC before the first reliable 18x.</div=
>

<div>=A0</div>
<div><font color=3D"#ff0000">[Eric]: Surely the UAS is allowed to send anot=
her reliable response until the first one is acknowledged, but reliable 199=
 will be useful if the first 18x is unreliable, or </font></div>
<div><font color=3D"#ff0000">the first 18x has been acknwledged.</font></di=
v>
<div><font color=3D"#ff0000">It&#39;s another case different from the above=
 one=A0 whose INVITE contains &quot;require: 100rel&quot;.</font></div>
<div><br><br>&gt;Can anyone show me how to sort all comments of a topic by =
time?<br><br>Unless it can be done by sorting the e-mails by subject it is =
kinda<br>difficult.<br><br>Regards,<br><br>Christer<br><br><br>Regards,<br>
<br>Eric.wang<br><br><br><br><br><br><br><br>=A0</div>

--000e0cd24b9ec22e7f0463d34a79--

From christer.holmberg@ericsson.com  Thu Feb 26 11:03:46 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Eric wang" <eric.wangxr@gmail.com>
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Hi,

>I still have some difficult in using IETF and cannot find the whole
sorted comments about this draft, so i have some doubt about this draft.
>
>   1.   "If a forking proxy receives a reliably sent 199 response for a
dialog, for which the proxy has previously generated and sent a 199
response, the proxy MUST forward the 199 response."
>
>     Does it describe the case below? Although P1 have sent a 199
response, P1 havs to  forword the send reliably 199 too.Or  is the
first 199 mistaked for 18x?
>
>          UAC               P1                  UAS_2
>            --- INVITE ------>
>                             --- INVITE (leg 2) ->
>            <-- 199(leg 2) --
>                             <-- 199 (leg 2) -----
>            <-- 199(leg 2) --

>I think it shall be 199, as currently written.
>=20
>[Eric]: If  this is 199, then Is there a special purpose to send two
199 in the same dialog?
>I think that one 199 is enough.
>And  the first 199 may be reliable too, that will make it a little
difficult to send the second 199.
=20
The second 199 could include information which is to be forwarded to the
UAC.


>2.  "10.  Usage with 100rel
>
>   When a 199 Early Dialog Terminated provisional response is sent by a
UAS, since the provisional response is only used for information
purpose, the UAS SHOULD send it unreliably even if the 100rel
>   option tag [RFC3262] is present in the Require header of the
associated request."
>
>
>I have seen a comment on this question,but still not understood about
>it. If the INVITE has a Require tag "Require: 100rel",does the UAS
still
>use unreliable 199 response?
>
>That is the recommendation, yes. The reasons is that we want to keep
199
>as "lightweight" as possible, without requireing re-transmissions and
>PRACKs.
>=20
>[Eric]: But reliable 199 have more advantage, and It is worth to use
the reliable 199,I think.

I don't know what that advantage would be, compared to having to send
PRACKs etc. This has been discussed quite much, so I would really need
some good justification to change it now.

Also, the draft doesn't forbid you to send the 199 reliably. It's only a
SHOULD.


>If 199 is reliable, there is one more advantage. If the 199 arrives
>before the first 18x response, UAC can discard the first 199 and
process
>it until UAC receives the first 18x response that has the same
>to-tag as 199, as the reliable 199 should be re-transmited until
>received PRACK.
>
>If the INVITE contains "Require: 100rel", the first 18x must also be
>sent reliably. And, I don't think the UAS is allowed to send another
>reliable response until the first one is acknowledged, so I don't think
>a reliable 199 would reach the UAC before the first reliable 18x.
>=20
>[Eric]: Surely the UAS is allowed to send another reliable response
until the first one is acknowledged,

If I remember correctly, the FIRST reliable response must be acknowleded
before the next reliable response is sent. But, we can double check in
the PRACK spec.

>but reliable 199 will be useful if the first 18x is unreliable, or the
first 18x has been acknwledged. It's another case different from the
above one  whose INVITE contains "require: 100rel".

If 100rel is required the 18x cannot be unreliable.

Regards,

Christer







=20

From drage@alcatel-lucent.com  Thu Feb 26 16:07:10 2009
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(As WG chair)

For information of the WG see below.

Please however try and make the existing deadline if you can, as the deadli=
nes are set to give people time to read the documents and make comments on =
them. Particularly for new author documents, we expect to see discussion on=
 the list identifying issues for discussion, and there are no automatic fir=
st time agenda slots! Therefore the two extra days you take may be the two =
days discussion that would have given you an agenda slot!

When submitting documents, please do take note of the new IPR requirements.=
 In particular make sure that:=20

-	if your company rather than you controls the IPR of your work, that they =
are aware of the new submission rules in RFC 5378.

-	if you incorporate material from other authors, that those authors should=
 be in accordance with the copyright submission statements that you are mak=
ing.

regards

Keith

-----Original Message-----
From: wgchairs-bounces@ietf.org [mailto:wgchairs-bounces@ietf.org] On Behal=
f Of Alexa Morris
Sent: Thursday, February 26, 2009 7:28 PM
To: ietf-announce@ietf.org; ietf@ietf.org; wgchairs@ietf.org
Subject: Initial Version I-D Submission Deadline Extended to March 4, 2009

The IESG has extended the deadline for initial version (00) submissions of =
Internet Drafts by 2 days (48 hours). The new deadline is March 4, 2009 at =
1700 Pacific (March 5, 2009 at  0100 UTC / GMT) and the extension is for IE=
TF 74 only.  The deadline has been extended due to the copyright legend tex=
t alternative being recently finalized, approved and implemented.

Please note that the date for updated I-D versions has NOT been extended, a=
nd is still March 9, 2009.

Regards,
Alexa

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Hi All,

 

I am currently working on the interworking between SIP and H323.

 

I had difficult in finding the status of SIP supporting dual video (e.g.
H.239 from a H.323 endpoint).

Are there any drafts or standards working on it? 

 

Also with the text chatting in a SIP call, if anyone can give me the RFC
number that would be great.

 

Thanks,

Yang


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Hi All,<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>I am currently working on the interworking =
between
SIP and H323.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>I had difficult in finding the status of SIP
supporting dual video (e.g. H.239 from a H.323 =
endpoint).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Are there any drafts or standards working on =
it? <o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Also with the text chatting in a SIP call, if =
anyone
can give me the RFC number that would be =
great.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New"'>Yang<o:p></o:p></span></font></p>

</div>

</body>

</html>

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*allowing* (much less *requiring*) the 199 to be reliable introduces
nasty problems. The 199 is only an optimization, so having it be
unreliable is ok IMO.

The problem is that if a proxy sends the 199, then the recipient of the
PRACK should be the proxy. But the PRACK is an in-dialog message, so it
must be addressed to the Contact of the UAS.

If the proxy sends a reliable 199, and the PRACK is addressed to the
UAS, the UAS will be very surprised, since it has not send a reliable
provisional, and is so not expecting a PRACK. In fact, it has sent a
final response, has presumably already received the ACK, and so is
expecting nothing.

Very ugly.

	Paul

wang.libo@zte.com.cn wrote:
> 
> Hi,
> 
> And there still two questions left.
> 
> 1.  Is the 199 should be reliable or unreliable?
>     I think that 199 should be reliable if possible.
> 
> 
>    First, If a client receives an unreliable 199 response on a dialog 
> which has
>    not previously been created (this can happen if a 199 response
>    reaches the client before a 18x response) the client SHALL discard
>    the 199 responses.
> 
>    The figure below shows an example.The 180 is sent first but arrives 
> later than 199.
> If the 199 is reliable, the proxy should retransmit 199 (step 4),and 
> then the retransmitted
> 199 will be accepted by UAC,and the early dialog will be teminated.
> 
> UAC             P
>    1. INVITE
> --------------->
>           2. 180
> <----- \/-------
>        /\ 3. 199
> <-----/  \------
>           4. 199(retransmitted)
> <---------------
> 
>    second, According to practical use, 199 can be intended to teminate 
> one early dialog and release
> resources associated with the specific early dialog, so the cost spent 
> on reliable 199 is worthy.
> If the 199 cannot be sent reliable,then we should send it unreliable.
>  
> 2. In your last letter, you said "the second 199 could include 
> information which is to be forwarded to the
> UAC", then do you mean, the early dialog is still alive after the first 
> 199 is accepted?
> 
> 
> Also, In my last letter, I miss a "NOT" by mistake.
> It should be
> [Eric]: Surely the UAS is NOT allowed to send another reliable response
> until the first one is acknowledged...
> 
> Regards,
> Eric wang
> 
> 
> 
> 
> *"Christer Holmberg" <christer.holmberg@ericsson.com>*
> 
> 2009-02-27 03:04
> 
> 	
> ÊÕ¼þÈË
> 	"Eric wang" <eric.wangxr@gmail.com>
> ³­ËÍ
> 	<sip@ietf.org>, <wang.libo@zte.com.cn>
> Ö÷Ìâ
> 	RE: Questions about "draft-ietf-sip-199-05"
> 
> 
> 	
> 
> 
> 
> 
> 
> Hi,
> 
>  >I still have some difficult in using IETF and cannot find the whole
> sorted comments about this draft, so i have some doubt about this draft.
>  >
>  >   1.   "If a forking proxy receives a reliably sent 199 response for a
> dialog, for which the proxy has previously generated and sent a 199
> response, the proxy MUST forward the 199 response."
>  >
>  >     Does it describe the case below? Although P1 have sent a 199
> response, P1 havs to  forword the send reliably 199 too.Or  is the
> first 199 mistaked for 18x?
>  >
>  >          UAC               P1                  UAS_2
>  >            --- INVITE ------>
>  >                             --- INVITE (leg 2) ->
>  >            <-- 199(leg 2) --
>  >                             <-- 199 (leg 2) -----
>  >            <-- 199(leg 2) --
> 
>  >I think it shall be 199, as currently written.
>  >
>  >[Eric]: If  this is 199, then Is there a special purpose to send two
> 199 in the same dialog?
>  >I think that one 199 is enough.
>  >And  the first 199 may be reliable too, that will make it a little
> difficult to send the second 199.
> 
> The second 199 could include information which is to be forwarded to the
> UAC.
> 
> 
>  >2.  "10.  Usage with 100rel
>  >
>  >   When a 199 Early Dialog Terminated provisional response is sent by a
> UAS, since the provisional response is only used for information
> purpose, the UAS SHOULD send it unreliably even if the 100rel
>  >   option tag [RFC3262] is present in the Require header of the
> associated request."
>  >
>  >
>  >I have seen a comment on this question,but still not understood about
>  >it. If the INVITE has a Require tag "Require: 100rel",does the UAS
> still
>  >use unreliable 199 response?
>  >
>  >That is the recommendation, yes. The reasons is that we want to keep
> 199
>  >as "lightweight" as possible, without requireing re-transmissions and
>  >PRACKs.
>  >
>  >[Eric]: But reliable 199 have more advantage, and It is worth to use
> the reliable 199,I think.
> 
> I don't know what that advantage would be, compared to having to send
> PRACKs etc. This has been discussed quite much, so I would really need
> some good justification to change it now.
> 
> Also, the draft doesn't forbid you to send the 199 reliably. It's only a
> SHOULD.
> 
> 
>  >If 199 is reliable, there is one more advantage. If the 199 arrives
>  >before the first 18x response, UAC can discard the first 199 and
> process
>  >it until UAC receives the first 18x response that has the same
>  >to-tag as 199, as the reliable 199 should be re-transmited until
>  >received PRACK.
>  >
>  >If the INVITE contains "Require: 100rel", the first 18x must also be
>  >sent reliably. And, I don't think the UAS is allowed to send another
>  >reliable response until the first one is acknowledged, so I don't think
>  >a reliable 199 would reach the UAC before the first reliable 18x.
>  >
>  >[Eric]: Surely the UAS is allowed to send another reliable response
> until the first one is acknowledged,
> 
> If I remember correctly, the FIRST reliable response must be acknowleded
> before the next reliable response is sent. But, we can double check in
> the PRACK spec.
> 
>  >but reliable 199 will be useful if the first 18x is unreliable, or the
> first 18x has been acknwledged. It's another case different from the
> above one  whose INVITE contains "require: 100rel".
> 
> If 100rel is required the 18x cannot be unreliable.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
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From eric.wangxr@gmail.com  Fri Feb 27 02:48:59 2009
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From: Eric wang <eric.wangxr@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [Sip] =?gb2312?b?tPC4tDogUkU6IFF1ZXN0aW9ucyBhYm91dCAiZHJhZnQtaWV0?= =?gb2312?b?Zi1zaXAtMTk5LTA1Ig==?=
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--000e0cd2e25867418b0463e438bd
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hi,

   If the UAC requires 100rel ,then what should the proxy do if it receives
a no-2xx
final response after forking?

Regards,
Eric



2009/2/27, Christer Holmberg <christer.holmberg@ericsson.com>:
>
>  Hi,
>
> >Practically, when Call forward and fork are used in the same system,
> there are
> >lots of early dialogs that should be eliminated by 199 existed.
> >So I think that it's useful to let 199 reliable.
> >And we just added fork-service to our system ,and find 199 is really
> useful, and
> >if the 199 is missing, problems will happen.
>
> I'm glad you think 199 is useful :)
>
> But, if a proxy would have to terminate PRACKs etc it would not be a prox=
y
> anymore - it would be a B2BUA.
>
> So, if you really want your network to send 199 reliably, I guess you cou=
ld
> use a B2BUA instead of a proxy.
>
> >It sounds good that there is a way to prevent PRACK addressed to UAS,but
> i don't know  how to do now.
>
> The only way to prevent it is by the proxy not sending the 199 relaible.
>
> >ps:
> >RFC3261 *16.7 Response Processing*
> >Since a proxy may not insert a tag into the To header field of
> >a 1xx response to a request that did not contain one, it cannot
> >issue non-100 provisional responses on its own.
>
> Yes, but I we have agreed that a proxy is allowed to send 199.
>
> But, the proxy is not going to generate new To header tag values for the
> 199. It will use whatever tags that have already been created for the ear=
ly
> dialogs.
>
> Regards,
>
> Christer
>
>
>
>
>
>
>
>
>
>
> *allowing* (much less *requiring*) the 199 to be reliable introduces
> nasty problems. The 199 is only an optimization, so having it be
> unreliable is ok IMO.
>
> The problem is that if a proxy sends the 199, then the recipient of the
> PRACK should be the proxy. But the PRACK is an in-dialog message, so it
> must be addressed to the Contact of the UAS.
>
> If the proxy sends a reliable 199, and the PRACK is addressed to the
> UAS, the UAS will be very surprised, since it has not send a reliable
> provisional, and is so not expecting a PRACK. In fact, it has sent a
> final response, has presumably already received the ACK, and so is
> expecting nothing.
>
> Very ugly.
>
>                 Paul
>
> wang.libo@zte.com.cn wrote:
> >
> > Hi,
> >
> > And there still two questions left.
> >
> > 1.  Is the 199 should be reliable or unreliable?
> >     I think that 199 should be reliable if possible.
> >
> >
> >    First, If a client receives an unreliable 199 response on a dialog
> > which has
> >    not previously been created (this can happen if a 199 response
> >    reaches the client before a 18x response) the client SHALL discard
> >    the 199 responses.
> >
> >    The figure below shows an example.The 180 is sent first but arrives
> > later than 199.
> > If the 199 is reliable, the proxy should retransmit 199 (step 4),and
> > then the retransmitted
> > 199 will be accepted by UAC,and the early dialog will be teminated.
> >
> > UAC             P
> >    1. INVITE
> > --------------->
> >           2. 180
> > <----- \/-------
> >        /\ 3. 199
> > <-----/  \------
> >           4. 199(retransmitted)
> > <---------------
> >
> >    second, According to practical use, 199 can be intended to teminate
> > one early dialog and release
> > resources associated with the specific early dialog, so the cost spent
> > on reliable 199 is worthy.
> > If the 199 cannot be sent reliable,then we should send it unreliable.
> >
> > 2. In your last letter, you said "the second 199 could include
> > information which is to be forwarded to the
> > UAC", then do you mean, the early dialog is still alive after the first
> > 199 is accepted?
> >
> >
> > Also, In my last letter, I miss a "NOT" by mistake.
> > It should be
> > [Eric]: Surely the UAS is NOT allowed to send another reliable response
> > until the first one is acknowledged...
> >
> > Regards,
> > Eric wang
> >
> >
> >
> >
> > *"Christer Holmberg" <christer.holmberg@ericsson.com>*
> >
> > 2009-02-27 03:04
> >
> >
> > =CA=D5=BC=FE=C8=CB
> >                  "Eric wang" <eric.wangxr@gmail.com>
> > =B3=AD=CB=CD
> >                  <sip@ietf.org>, <wang.libo@zte.com.cn>
> > =D6=F7=CC=E2
> >                  RE: Questions about "draft-ietf-sip-199-05"
> >
> >
> >
> >
> >
> >
> >
> >
> > Hi,
> >
> >  >I still have some difficult in using IETF and cannot find the whole
> > sorted comments about this draft, so i have some doubt about this draft=
.
> >  >
> >  >   1.   "If a forking proxy receives a reliably sent 199 response for=
 a
> > dialog, for which the proxy has previously generated and sent a 199
> > response, the proxy MUST forward the 199 response."
> >  >
> >  >     Does it describe the case below? Although P1 have sent a 199
> > response, P1 havs to  forword the send reliably 199 too.Or  is the
> > first 199 mistaked for 18x?
> >  >
> >  >          UAC               P1                  UAS_2
> >  >            --- INVITE ------>
> >  >                             --- INVITE (leg 2) ->
> >  >            <-- 199(leg 2) --
> >  >                             <-- 199 (leg 2) -----
> >  >            <-- 199(leg 2) --
> >
> >  >I think it shall be 199, as currently written.
> >  >
> >  >[Eric]: If  this is 199, then Is there a special purpose to send two
> > 199 in the same dialog?
> >  >I think that one 199 is enough.
> >  >And  the first 199 may be reliable too, that will make it a little
> > difficult to send the second 199.
> >
> > The second 199 could include information which is to be forwarded to th=
e
> > UAC.
> >
> >
> >  >2.  "10.  Usage with 100rel
> >  >
> >  >   When a 199 Early Dialog Terminated provisional response is sent by=
 a
> > UAS, since the provisional response is only used for information
> > purpose, the UAS SHOULD send it unreliably even if the 100rel
> >  >   option tag [RFC3262] is present in the Require header of the
> > associated request."
> >  >
> >  >
> >  >I have seen a comment on this question,but still not understood about
> >  >it. If the INVITE has a Require tag "Require: 100rel",does the UAS
> > still
> >  >use unreliable 199 response?
> >  >
> >  >That is the recommendation, yes. The reasons is that we want to keep
> > 199
> >  >as "lightweight" as possible, without requireing re-transmissions and
> >  >PRACKs.
> >  >
> >  >[Eric]: But reliable 199 have more advantage, and It is worth to use
> > the reliable 199,I think.
> >
> > I don't know what that advantage would be, compared to having to send
> > PRACKs etc. This has been discussed quite much, so I would really need
> > some good justification to change it now.
> >
> > Also, the draft doesn't forbid you to send the 199 reliably. It's only =
a
> > SHOULD.
> >
> >
> >  >If 199 is reliable, there is one more advantage. If the 199 arrives
> >  >before the first 18x response, UAC can discard the first 199 and
> > process
> >  >it until UAC receives the first 18x response that has the same
> >  >to-tag as 199, as the reliable 199 should be re-transmited until
> >  >received PRACK.
> >  >
> >  >If the INVITE contains "Require: 100rel", the first 18x must also be
> >  >sent reliably. And, I don't think the UAS is allowed to send another
> >  >reliable response until the first one is acknowledged, so I don't thi=
nk
> >  >a reliable 199 would reach the UAC before the first reliable 18x.
> >  >
> >  >[Eric]: Surely the UAS is allowed to send another reliable response
> > until the first one is acknowledged,
> >
> > If I remember correctly, the FIRST reliable response must be acknowlede=
d
> > before the next reliable response is sent. But, we can double check in
> > the PRACK spec.
> >
> >  >but reliable 199 will be useful if the first 18x is unreliable, or th=
e
> > first 18x has been acknwledged. It's another case different from the
> > above one  whose INVITE contains "require: 100rel".
> >
> > If 100rel is required the 18x cannot be unreliable.
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > --------------------------------------------------------
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is
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d
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> > This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they are
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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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Content-Type: text/html; charset=GB2312
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<div>hi,</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; If the UAC requires 100rel ,then what should the proxy do=
 if it receives a no-2xx</div>
<div>final response after forking?</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>Eric</div>
<div><br><br>&nbsp;</div>
<div><span class=3D"gmail_quote">2009/2/27, Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;:</span>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><span class=3D"q">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">Hi,</font></span></div>
<div><font face=3D"sans-serif" size=3D"2">&nbsp;&nbsp;</font><br><font face=
=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font color=3D"#0=
00000">&gt;</font></font></span>Practically, when Call forward and fork are=
 used in the same system, there are&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>lots of early dialogs that should=
 be eliminated by 199 existed.</font></font>&nbsp;<br><font face=3D"Arial">=
<font size=3D"2"><span><font color=3D"#0000ff"><font face=3D"" color=3D"#00=
0000" size=3D"3">&gt;</font></font></span>So I think that it&#39;s useful t=
o let 199 reliable.&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>And we just added fork-service to=
 our system ,and find 199 is really useful, and</font></font>&nbsp;<br><fon=
t face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font face=
=3D"" color=3D"#000000" size=3D"3">&gt;</font></font></span>if the 199 is m=
issing, problems will happen.</font></font>&nbsp;<span><font face=3D"Arial"=
 color=3D"#0000ff" size=3D"2">&nbsp;</font></span></div>

<div><span></span>&nbsp;</div></span>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">I&#39;m glad y=
ou think 199 is useful :)</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">But, if a prox=
y would have to terminate PRACKs etc it would not be a proxy anymore - it w=
ould be a B2BUA.</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">So, if you rea=
lly want your network to send 199 reliably, I guess you could use a B2BUA i=
nstead of a proxy.</font></span></div><span class=3D"q"><font face=3D"Arial=
" color=3D"#0000ff" size=3D"2"></font>
<div><br><font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff=
">&gt;</font></span>It sounds good that there is a way to prevent PRACK add=
ressed to UAS,but i don&#39;t know<span><font color=3D"#0000ff">&nbsp; </fo=
nt></span>how to do now.<span><font color=3D"#0000ff">&nbsp;</font></span><=
/font></font></div>

<div><font face=3D"Arial"><font size=3D"2"><span></span></font></font>&nbsp=
;</div></span>
<div><font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff">Th=
e only way to prevent it is by the proxy not sending the 199&nbsp;relaible.=
</font>&nbsp;</span></font></font>&nbsp;<span class=3D"q"><br><font face=3D=
"sans-serif" size=3D"2">&nbsp;&nbsp;&nbsp;</font>&nbsp;<br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font f=
ace=3D"Times New Roman" color=3D"#000000" size=3D"3">&gt;</font></font></sp=
an>ps:</font></font>&nbsp;<br><font size=3D"2"><font face=3D"Arial"><span><=
font color=3D"#0000ff"><font face=3D"Times New Roman" color=3D"#000000" siz=
e=3D"3">&gt;</font></font></span>RFC3261 </font><font face=3D"Courier New" =
color=3D"#002041"><b>16.7 Response Processing</b></font><font face=3D"sans-=
serif">&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>Since a proxy may not insert a ta=
g into the To header field of</font></font>&nbsp;<br><font face=3D"Arial"><=
font size=3D"2"><span><font color=3D"#0000ff">&gt;</font></span>a 1xx respo=
nse to a request that did not contain one, <font color=3D"blue">it cannot</=
font></font></font>&nbsp;<br>
<font face=3D"Arial"><font color=3D"#0000ff"><font size=3D"2"><span><font f=
ace=3D"Times New Roman" color=3D"#000000" size=3D"3">&gt;</font></span>issu=
e non-100 provisional responses on its own.</font></font></font>&nbsp;<span=
><font face=3D"Arial" color=3D"#0000ff" size=3D"2">&nbsp;</font></span></sp=
an></div>

<div><span></span>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Yes, but I&nbs=
p;we have agreed that a proxy is allowed to send 199.</font></span></div>
<div><span></span>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">But, the proxy=
 is not going to generate new To header tag values for the 199. It will use=
 whatever tags that have already&nbsp;been created for the&nbsp;early dialo=
gs.</font>&nbsp;</span></div>

<div><span></span><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">&=
nbsp;</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Regards,</font=
></span></div>
<div><span></span>&nbsp;</div><span class=3D"sg">
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Christer</font=
></span></div></span>
<div><span class=3D"e" id=3D"q_11fb6678b166112b_8">
<div><span>&nbsp;</span><br><span><font face=3D"Arial" color=3D"#0000ff" si=
ze=3D"2">&nbsp;</font></span></div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span>&nbsp;</span><br><br><tt><font size=3D"2">*allowing* (much less =
*requiring*) the 199 to be reliable introduces<br>nasty problems. The 199 i=
s only an optimization, so having it be<br>unreliable is ok IMO.<br><br>The=
 problem is that if a proxy sends the 199, then the recipient of the<br>
PRACK should be the proxy. But the PRACK is an in-dialog message, so it<br>=
must be addressed to the Contact of the UAS.<br><br>If the proxy sends a re=
liable 199, and the PRACK is addressed to the<br>UAS, the UAS will be very =
surprised, since it has not send a reliable<br>
provisional, and is so not expecting a PRACK. In fact, it has sent a<br>fin=
al response, has presumably already received the ACK, and so is<br>expectin=
g nothing.<br><br>Very ugly.<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; Paul<br><br><a onclick=3D"return top.js.OpenExtLink(window=
,event,this)" href=3D"mailto:wang.libo@zte.com.cn" target=3D"_blank">wang.l=
ibo@zte.com.cn</a> wrote:<br>
&gt; <br>&gt; Hi,<br>&gt; <br>&gt; And there still two questions left.<br>&=
gt; <br>&gt; 1. &nbsp;Is the 199 should be reliable or unreliable?<br>&gt; =
&nbsp; &nbsp; I think that 199 should be reliable if possible.<br>&gt; <br>=
&gt; <br>&gt; &nbsp; &nbsp;First, If a client receives an unreliable 199 re=
sponse on a dialog <br>
&gt; which has<br>&gt; &nbsp; &nbsp;not previously been created (this can h=
appen if a 199 response<br>&gt; &nbsp; &nbsp;reaches the client before a 18=
x response) the client SHALL discard<br>&gt; &nbsp; &nbsp;the 199 responses=
.<br>&gt; <br>&gt; &nbsp; &nbsp;The figure below shows an example.The 180 i=
s sent first but arrives <br>
&gt; later than 199.<br>&gt; If the 199 is reliable, the proxy should retra=
nsmit 199 (step 4),and <br>&gt; then the retransmitted<br>&gt; 199 will be =
accepted by UAC,and the early dialog will be teminated.<br>&gt; <br>&gt; UA=
C &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; P<br>
&gt; &nbsp; &nbsp;1. INVITE<br>&gt; ---------------&gt;<br>&gt; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; 2. 180<br>&gt; &lt;----- \/-------<br>&gt; &nbsp; &=
nbsp; &nbsp; &nbsp;/\ 3. 199<br>&gt; &lt;-----/ &nbsp;\------<br>&gt; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; 4. 199(retransmitted)<br>&gt; &lt;-----------=
----<br>
&gt; <br>&gt; &nbsp; &nbsp;second, According to practical use, 199 can be i=
ntended to teminate <br>&gt; one early dialog and release<br>&gt; resources=
 associated with the specific early dialog, so the cost spent <br>&gt; on r=
eliable 199 is worthy.<br>
&gt; If the 199 cannot be sent reliable,then we should send it unreliable.<=
br>&gt; &nbsp;<br>&gt; 2. In your last letter, you said &quot;the second 19=
9 could include <br>&gt; information which is to be forwarded to the<br>&gt=
; UAC&quot;, then do you mean, the early dialog is still alive after the fi=
rst <br>
&gt; 199 is accepted?<br>&gt; <br>&gt; <br>&gt; Also, In my last letter, I =
miss a &quot;NOT&quot; by mistake.<br>&gt; It should be<br>&gt; [Eric]: Sur=
ely the UAS is NOT allowed to send another reliable response<br>&gt; until =
the first one is acknowledged...<br>
&gt; <br>&gt; Regards,<br>&gt; Eric wang<br>&gt; <br>&gt; <br>&gt; <br>&gt;=
 <br>&gt; *&quot;Christer Holmberg&quot; &lt;<a onclick=3D"return top.js.Op=
enExtLink(window,event,this)" href=3D"mailto:christer.holmberg@ericsson.com=
" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;*<br>
&gt; <br>&gt; 2009-02-27 03:04<br>&gt; <br>&gt; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<br>&gt; =CA=D5=BC=FE=C8=CB<br>&gt; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&quot;Eric wang&q=
uot; &lt;<a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=
=3D"mailto:eric.wangxr@gmail.com" target=3D"_blank">eric.wangxr@gmail.com</=
a>&gt;<br>
&gt; =B3=AD=CB=CD<br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;&lt;<a onclick=3D"return top.js.OpenExtLink(window,event,this)=
" href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>&gt;, &lt;=
<a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=3D"mailto:=
wang.libo@zte.com.cn" target=3D"_blank">wang.libo@zte.com.cn</a>&gt;<br>
&gt; =D6=F7=CC=E2<br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;RE: Questions about &quot;draft-ietf-sip-199-05&quot;<br>&gt; =
<br>&gt; <br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;<br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; Hi,<br>&gt; <br>=
&gt; &nbsp;&gt;I still have some difficult in using IETF and cannot find th=
e whole<br>
&gt; sorted comments about this draft, so i have some doubt about this draf=
t.<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt; &nbsp; 1. &nbsp; &quot;If a forkin=
g proxy receives a reliably sent 199 response for a<br>&gt; dialog, for whi=
ch the proxy has previously generated and sent a 199<br>
&gt; response, the proxy MUST forward the 199 response.&quot;<br>&gt; &nbsp=
;&gt;<br>&gt; &nbsp;&gt; &nbsp; &nbsp; Does it describe the case below? Alt=
hough P1 have sent a 199<br>&gt; response, P1 havs to &nbsp;forword the sen=
d reliably 199 too.Or &nbsp;is the<br>
&gt; first 199 mistaked for 18x?<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp;UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; P1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;UA=
S_2<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;--- INVITE =
------&gt;<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; --- INVITE (leg 2) -&g=
t;<br>
&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- 199(leg 2) =
--<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;-- 199 (leg 2) -----<br>&g=
t; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- 199(leg 2) --=
<br>&gt; <br>&gt; &nbsp;&gt;I think it shall be 199, as currently written.<=
br>
&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;[Eric]: If &nbsp;this is 199, then Is the=
re a special purpose to send two<br>&gt; 199 in the same dialog?<br>&gt; &n=
bsp;&gt;I think that one 199 is enough.<br>&gt; &nbsp;&gt;And &nbsp;the fir=
st 199 may be reliable too, that will make it a little<br>
&gt; difficult to send the second 199.<br>&gt; <br>&gt; The second 199 coul=
d include information which is to be forwarded to the<br>&gt; UAC.<br>&gt; =
<br>&gt; <br>&gt; &nbsp;&gt;2. &nbsp;&quot;10. &nbsp;Usage with 100rel<br>&=
gt; &nbsp;&gt;<br>
&gt; &nbsp;&gt; &nbsp; When a 199 Early Dialog Terminated provisional respo=
nse is sent by a<br>&gt; UAS, since the provisional response is only used f=
or information<br>&gt; purpose, the UAS SHOULD send it unreliably even if t=
he 100rel<br>
&gt; &nbsp;&gt; &nbsp; option tag [RFC3262] is present in the Require heade=
r of the<br>&gt; associated request.&quot;<br>&gt; &nbsp;&gt;<br>&gt; &nbsp=
;&gt;<br>&gt; &nbsp;&gt;I have seen a comment on this question,but still no=
t understood about<br>
&gt; &nbsp;&gt;it. If the INVITE has a Require tag &quot;Require: 100rel&qu=
ot;,does the UAS<br>&gt; still<br>&gt; &nbsp;&gt;use unreliable 199 respons=
e?<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;That is the recommendation, yes. Th=
e reasons is that we want to keep<br>
&gt; 199<br>&gt; &nbsp;&gt;as &quot;lightweight&quot; as possible, without =
requireing re-transmissions and<br>&gt; &nbsp;&gt;PRACKs.<br>&gt; &nbsp;&gt=
;<br>&gt; &nbsp;&gt;[Eric]: But reliable 199 have more advantage, and It is=
 worth to use<br>
&gt; the reliable 199,I think.<br>&gt; <br>&gt; I don&#39;t know what that =
advantage would be, compared to having to send<br>&gt; PRACKs etc. This has=
 been discussed quite much, so I would really need<br>&gt; some good justif=
ication to change it now.<br>
&gt; <br>&gt; Also, the draft doesn&#39;t forbid you to send the 199 reliab=
ly. It&#39;s only a<br>&gt; SHOULD.<br>&gt; <br>&gt; <br>&gt; &nbsp;&gt;If =
199 is reliable, there is one more advantage. If the 199 arrives<br>&gt; &n=
bsp;&gt;before the first 18x response, UAC can discard the first 199 and<br=
>
&gt; process<br>&gt; &nbsp;&gt;it until UAC receives the first 18x response=
 that has the same<br>&gt; &nbsp;&gt;to-tag as 199, as the reliable 199 sho=
uld be re-transmited until<br>&gt; &nbsp;&gt;received PRACK.<br>&gt; &nbsp;=
&gt;<br>&gt; &nbsp;&gt;If the INVITE contains &quot;Require: 100rel&quot;, =
the first 18x must also be<br>
&gt; &nbsp;&gt;sent reliably. And, I don&#39;t think the UAS is allowed to =
send another<br>&gt; &nbsp;&gt;reliable response until the first one is ack=
nowledged, so I don&#39;t think<br>&gt; &nbsp;&gt;a reliable 199 would reac=
h the UAC before the first reliable 18x.<br>
&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;[Eric]: Surely the UAS is allowed to send=
 another reliable response<br>&gt; until the first one is acknowledged,<br>=
&gt; <br>&gt; If I remember correctly, the FIRST reliable response must be =
acknowleded<br>
&gt; before the next reliable response is sent. But, we can double check in=
<br>&gt; the PRACK spec.<br>&gt; <br>&gt; &nbsp;&gt;but reliable 199 will b=
e useful if the first 18x is unreliable, or the<br>&gt; first 18x has been =
acknwledged. It&#39;s another case different from the<br>
&gt; above one &nbsp;whose INVITE contains &quot;require: 100rel&quot;.<br>=
&gt; <br>&gt; If 100rel is required the 18x cannot be unreliable.<br>&gt; <=
br>&gt; Regards,<br>&gt; <br>&gt; Christer<br>&gt; <br>&gt; <br>&gt; <br>&g=
t; <br>
&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; ----------------=
----------------------------------------<br>&gt; ZTE Information Security N=
otice: The information contained in this mail is solely property of the sen=
der&#39;s organization. This mail communication is confidential. Recipients=
 named above are obligated to maintain secrecy and are not permitted to dis=
close the contents of this communication to others.<br>
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nded solely for the use of the individual or entity to whom they are addres=
sed. If you have received this email in error please notify the originator =
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event,this)" href=3D"https://www.ietf.org/mailman/listinfo/sip" target=3D"_=
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or NEW development of the core SIP Protocol<br>
&gt; Use <a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=
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 application of sip<br>
</font></tt><br><br>&nbsp;</div><pre>--------------------------------------=
------------------
ZTE&nbsp;Information&nbsp;Security&nbsp;Notice:&nbsp;The&nbsp;information&n=
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</pre></span></div></div><br>______________________________________________=
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event,this)" href=3D"https://www.ietf.org/mailman/listinfo/sip" target=3D"_=
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This list is for NEW development of the core SIP Protocol<br>Use <a onclick=
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Use <a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=3D"mai=
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--000e0cd2e25867418b0463e438bd--

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From eric.wangxr@gmail.com  Fri Feb 27 03:10:09 2009
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References: <49A7484B.3060407@cisco.com> <OF331E0406.F68A068E-ON4825756A.001684FC-4825756A.00197C73@zte.com.cn> <CA9998CD4A020D418654FCDEF4E707DF0B167F1E@esealmw113.eemea.ericsson.se> <898632750902270249o3213670au6823cc4ab6180c8c@mail.gmail.com> <CA9998CD4A020D418654FCDEF4E707DF0B167F42@esealmw113.eemea.ericsson.se>
Date: Fri, 27 Feb 2009 19:10:29 +0800
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From: Eric wang <eric.wangxr@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: sip@ietf.org, Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] =?gb2312?b?tPC4tDogUkU6IFF1ZXN0aW9ucyBhYm91dCAiZHJhZnQtaWV0?= =?gb2312?b?Zi1zaXAtMTk5LTA1Ig==?=
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hi,

 I don't think the 199 can be sent unreliably if UAC require 100rel.
As in RFC3262, It says.

 The UAS MUST send any non-100 provisional response reliably if the
   initial request contained a Require header field with the option tag
   100rel.  If the UAS is unwilling to do so, it MUST reject the initial
   request with a 420 (Bad Extension) and include an Unsupported header
   field containing the option tag 100rel.

Regards,
Eric


2009/2/27, Christer Holmberg <christer.holmberg@ericsson.com>:
>
>  Hi,
>
> >If the UAC requires 100rel ,then what should the proxy do if it receives
> a no-2xx   final response after forking?
>
> The proxy should send 199 unreliably.
>
> Regards,
>
> Christer
>
>
>
>
> 2009/2/27, Christer Holmberg <christer.holmberg@ericsson.com>:
>>
>>  Hi,
>>
>> >Practically, when Call forward and fork are used in the same system,
>> there are
>> >lots of early dialogs that should be eliminated by 199 existed.
>> >So I think that it's useful to let 199 reliable.
>> >And we just added fork-service to our system ,and find 199 is really
>> useful, and
>> >if the 199 is missing, problems will happen.
>>
>> I'm glad you think 199 is useful :)
>>
>> But, if a proxy would have to terminate PRACKs etc it would not be a pro=
xy
>> anymore - it would be a B2BUA.
>>
>> So, if you really want your network to send 199 reliably, I guess you
>> could use a B2BUA instead of a proxy.
>>
>> >It sounds good that there is a way to prevent PRACK addressed to UAS,bu=
t
>> i don't know  how to do now.
>>
>> The only way to prevent it is by the proxy not sending the 199 relaible.
>>
>>
>> >ps:
>> >RFC3261 *16.7 Response Processing*
>> >Since a proxy may not insert a tag into the To header field of
>> >a 1xx response to a request that did not contain one, it cannot
>> >issue non-100 provisional responses on its own.
>>
>> Yes, but I we have agreed that a proxy is allowed to send 199.
>>
>> But, the proxy is not going to generate new To header tag values for the
>> 199. It will use whatever tags that have already been created for the ea=
rly
>> dialogs.
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *allowing* (much less *requiring*) the 199 to be reliable introduces
>> nasty problems. The 199 is only an optimization, so having it be
>> unreliable is ok IMO.
>>
>> The problem is that if a proxy sends the 199, then the recipient of the
>> PRACK should be the proxy. But the PRACK is an in-dialog message, so it
>> must be addressed to the Contact of the UAS.
>>
>> If the proxy sends a reliable 199, and the PRACK is addressed to the
>> UAS, the UAS will be very surprised, since it has not send a reliable
>> provisional, and is so not expecting a PRACK. In fact, it has sent a
>> final response, has presumably already received the ACK, and so is
>> expecting nothing.
>>
>> Very ugly.
>>
>>                 Paul
>>
>> wang.libo@zte.com.cn wrote:
>> >
>> > Hi,
>> >
>> > And there still two questions left.
>> >
>> > 1.  Is the 199 should be reliable or unreliable?
>> >     I think that 199 should be reliable if possible.
>> >
>> >
>> >    First, If a client receives an unreliable 199 response on a dialog
>> > which has
>> >    not previously been created (this can happen if a 199 response
>> >    reaches the client before a 18x response) the client SHALL discard
>> >    the 199 responses.
>> >
>> >    The figure below shows an example.The 180 is sent first but arrives
>> > later than 199.
>> > If the 199 is reliable, the proxy should retransmit 199 (step 4),and
>> > then the retransmitted
>> > 199 will be accepted by UAC,and the early dialog will be teminated.
>> >
>> > UAC             P
>> >    1. INVITE
>> > --------------->
>> >           2. 180
>> > <----- \/-------
>> >        /\ 3. 199
>> > <-----/  \------
>> >           4. 199(retransmitted)
>> > <---------------
>> >
>> >    second, According to practical use, 199 can be intended to teminate
>> > one early dialog and release
>> > resources associated with the specific early dialog, so the cost spent
>> > on reliable 199 is worthy.
>> > If the 199 cannot be sent reliable,then we should send it unreliable.
>> >
>> > 2. In your last letter, you said "the second 199 could include
>> > information which is to be forwarded to the
>> > UAC", then do you mean, the early dialog is still alive after the firs=
t
>> > 199 is accepted?
>> >
>> >
>> > Also, In my last letter, I miss a "NOT" by mistake.
>> > It should be
>> > [Eric]: Surely the UAS is NOT allowed to send another reliable respons=
e
>> > until the first one is acknowledged...
>> >
>> > Regards,
>> > Eric wang
>> >
>> >
>> >
>> >
>> > *"Christer Holmberg" <christer.holmberg@ericsson.com>*
>> >
>> > 2009-02-27 03:04
>> >
>> >
>> > =CA=D5=BC=FE=C8=CB
>> >                  "Eric wang" <eric.wangxr@gmail.com>
>> > =B3=AD=CB=CD
>> >                  <sip@ietf.org>, <wang.libo@zte.com.cn>
>> > =D6=F7=CC=E2
>> >                  RE: Questions about "draft-ietf-sip-199-05"
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > Hi,
>> >
>> >  >I still have some difficult in using IETF and cannot find the whole
>> > sorted comments about this draft, so i have some doubt about this draf=
t.
>> >  >
>> >  >   1.   "If a forking proxy receives a reliably sent 199 response fo=
r
>> a
>> > dialog, for which the proxy has previously generated and sent a 199
>> > response, the proxy MUST forward the 199 response."
>> >  >
>> >  >     Does it describe the case below? Although P1 have sent a 199
>> > response, P1 havs to  forword the send reliably 199 too.Or  is the
>> > first 199 mistaked for 18x?
>> >  >
>> >  >          UAC               P1                  UAS_2
>> >  >            --- INVITE ------>
>> >  >                             --- INVITE (leg 2) ->
>> >  >            <-- 199(leg 2) --
>> >  >                             <-- 199 (leg 2) -----
>> >  >            <-- 199(leg 2) --
>> >
>> >  >I think it shall be 199, as currently written.
>> >  >
>> >  >[Eric]: If  this is 199, then Is there a special purpose to send two
>> > 199 in the same dialog?
>> >  >I think that one 199 is enough.
>> >  >And  the first 199 may be reliable too, that will make it a little
>> > difficult to send the second 199.
>> >
>> > The second 199 could include information which is to be forwarded to t=
he
>> > UAC.
>> >
>> >
>> >  >2.  "10.  Usage with 100rel
>> >  >
>> >  >   When a 199 Early Dialog Terminated provisional response is sent b=
y
>> a
>> > UAS, since the provisional response is only used for information
>> > purpose, the UAS SHOULD send it unreliably even if the 100rel
>> >  >   option tag [RFC3262] is present in the Require header of the
>> > associated request."
>> >  >
>> >  >
>> >  >I have seen a comment on this question,but still not understood abou=
t
>> >  >it. If the INVITE has a Require tag "Require: 100rel",does the UAS
>> > still
>> >  >use unreliable 199 response?
>> >  >
>> >  >That is the recommendation, yes. The reasons is that we want to keep
>> > 199
>> >  >as "lightweight" as possible, without requireing re-transmissions an=
d
>> >  >PRACKs.
>> >  >
>> >  >[Eric]: But reliable 199 have more advantage, and It is worth to use
>> > the reliable 199,I think.
>> >
>> > I don't know what that advantage would be, compared to having to send
>> > PRACKs etc. This has been discussed quite much, so I would really need
>> > some good justification to change it now.
>> >
>> > Also, the draft doesn't forbid you to send the 199 reliably. It's only=
 a
>> > SHOULD.
>> >
>> >
>> >  >If 199 is reliable, there is one more advantage. If the 199 arrives
>> >  >before the first 18x response, UAC can discard the first 199 and
>> > process
>> >  >it until UAC receives the first 18x response that has the same
>> >  >to-tag as 199, as the reliable 199 should be re-transmited until
>> >  >received PRACK.
>> >  >
>> >  >If the INVITE contains "Require: 100rel", the first 18x must also be
>> >  >sent reliably. And, I don't think the UAS is allowed to send another
>> >  >reliable response until the first one is acknowledged, so I don't
>> think
>> >  >a reliable 199 would reach the UAC before the first reliable 18x.
>> >  >
>> >  >[Eric]: Surely the UAS is allowed to send another reliable response
>> > until the first one is acknowledged,
>> >
>> > If I remember correctly, the FIRST reliable response must be acknowled=
ed
>> > before the next reliable response is sent. But, we can double check in
>> > the PRACK spec.
>> >
>> >  >but reliable 199 will be useful if the first 18x is unreliable, or t=
he
>> > first 18x has been acknwledged. It's another case different from the
>> > above one  whose INVITE contains "require: 100rel".
>> >
>> > If 100rel is required the 18x cannot be unreliable.
>> >
>> > Regards,
>> >
>> > Christer
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > --------------------------------------------------------
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<div>hi, </div>
<div>&nbsp; </div>
<div>&nbsp;I don&#39;t think the 199 can be sent unreliably if UAC require =
100rel.</div>
<div>As&nbsp;in RFC3262, It says.</div>
<div>&nbsp;&nbsp;</div>
<div>&nbsp;The UAS MUST send any non-100 provisional response reliably if t=
he<br>&nbsp;&nbsp; initial request contained a Require header field with th=
e option tag<br>&nbsp;&nbsp; 100rel.&nbsp; If the UAS is unwilling to do so=
, it MUST reject the initial<br>
&nbsp;&nbsp; request with a 420 (Bad Extension) and include an Unsupported =
header<br>&nbsp;&nbsp; field containing the option tag 100rel.<br><br>Regar=
ds,</div>
<div>Eric</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><span class=3D"gmail_quote">2009/2/27, Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;:</span>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><span class=3D"q">
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">Hi,</font></span></div>
<div><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">&gt;</font></s=
pan>If the UAC requires 100rel ,then what should the proxy do if it receive=
s a no-2xx<span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">&nbsp; &n=
bsp;</font></span>final response after forking?<span><font face=3D"Arial" c=
olor=3D"#0000ff" size=3D"2">&nbsp;</font></span></div>

<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div></span>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">The proxy&nbsp=
;should send 199 unreliably.</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Regards,</font=
></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div><span class=3D"sg">
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Christer</font=
>&nbsp;</span></div></span>
<div><span class=3D"e" id=3D"q_11fb7649e2b738d3_4">
<div><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font>&nbsp;</div>
<div><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font><br><br>&nbsp=
;</div>
<div><span class=3D"gmail_quote">2009/2/27, Christer Holmberg &lt;<a onclic=
k=3D"return top.js.OpenExtLink(window,event,this)" href=3D"mailto:christer.=
holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>=
&gt;:</span>=20
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><span>
<div dir=3D"ltr" align=3D"left"><span><font face=3D"Arial" color=3D"#0000ff=
" size=3D"2">Hi,</font></span></div>
<div><font face=3D"sans-serif" size=3D"2">&nbsp;&nbsp;</font><br><font face=
=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font color=3D"#0=
00000">&gt;</font></font></span>Practically, when Call forward and fork are=
 used in the same system, there are&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>lots of early dialogs that should=
 be eliminated by 199 existed.</font></font>&nbsp;<br><font face=3D"Arial">=
<font size=3D"2"><span><font color=3D"#0000ff"><font face=3D"" color=3D"#00=
0000" size=3D"3">&gt;</font></font></span>So I think that it&#39;s useful t=
o let 199 reliable.&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>And we just added fork-service to=
 our system ,and find 199 is really useful, and</font></font>&nbsp;<br><fon=
t face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font face=
=3D"" color=3D"#000000" size=3D"3">&gt;</font></font></span>if the 199 is m=
issing, problems will happen.</font></font>&nbsp;<span><font face=3D"Arial"=
 color=3D"#0000ff" size=3D"2">&nbsp;</font></span></div>

<div><span></span>&nbsp;</div></span>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">I&#39;m glad y=
ou think 199 is useful :)</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">But, if a prox=
y would have to terminate PRACKs etc it would not be a proxy anymore - it w=
ould be a B2BUA.</font></span></div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"></font></span>=
&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">So, if you rea=
lly want your network to send 199 reliably, I guess you could use a B2BUA i=
nstead of a proxy.</font></span></div><span><font face=3D"Arial" color=3D"#=
0000ff" size=3D"2"></font>
<div><br><font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff=
">&gt;</font></span>It sounds good that there is a way to prevent PRACK add=
ressed to UAS,but i don&#39;t know<span><font color=3D"#0000ff">&nbsp; </fo=
nt></span>how to do now.<span><font color=3D"#0000ff">&nbsp;</font></span><=
/font></font></div>

<div><font face=3D"Arial"><font size=3D"2"><span></span></font></font>&nbsp=
;</div></span>
<div><font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff">Th=
e only way to prevent it is by the proxy not sending the 199&nbsp;relaible.=
</font>&nbsp;</span></font></font>&nbsp;<span><br><font face=3D"sans-serif"=
 size=3D"2">&nbsp;&nbsp;&nbsp;</font>&nbsp;<br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font f=
ace=3D"Times New Roman" color=3D"#000000" size=3D"3">&gt;</font></font></sp=
an>ps:</font></font>&nbsp;<br><font size=3D"2"><font face=3D"Arial"><span><=
font color=3D"#0000ff"><font face=3D"Times New Roman" color=3D"#000000" siz=
e=3D"3">&gt;</font></font></span>RFC3261 </font><font face=3D"Courier New" =
color=3D"#002041"><b>16.7 Response Processing</b></font><font face=3D"sans-=
serif">&nbsp;</font></font><br>
<font face=3D"Arial"><font size=3D"2"><span><font color=3D"#0000ff"><font c=
olor=3D"#000000">&gt;</font></font></span>Since a proxy may not insert a ta=
g into the To header field of</font></font>&nbsp;<br><font face=3D"Arial"><=
font size=3D"2"><span><font color=3D"#0000ff">&gt;</font></span>a 1xx respo=
nse to a request that did not contain one, <font color=3D"blue">it cannot</=
font></font></font>&nbsp;<br>
<font face=3D"Arial"><font color=3D"#0000ff"><font size=3D"2"><span><font f=
ace=3D"Times New Roman" color=3D"#000000" size=3D"3">&gt;</font></span>issu=
e non-100 provisional responses on its own.</font></font></font>&nbsp;<span=
><font face=3D"Arial" color=3D"#0000ff" size=3D"2">&nbsp;</font></span></sp=
an></div>

<div><span></span>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Yes, but I&nbs=
p;we have agreed that a proxy is allowed to send 199.</font></span></div>
<div><span></span>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">But, the proxy=
 is not going to generate new To header tag values for the 199. It will use=
 whatever tags that have already&nbsp;been created for the&nbsp;early dialo=
gs.</font>&nbsp;</span></div>

<div><span></span><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2"><=
/font></span>&nbsp;</div>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Regards,</font=
></span></div>
<div><span></span>&nbsp;</div><span>
<div><span><font face=3D"Arial" color=3D"#0000ff" size=3D"2">Christer</font=
></span></div></span>
<div><span>
<div><span></span><br><span><font face=3D"Arial" color=3D"#0000ff" size=3D"=
2">&nbsp;</font></span></div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span>&nbsp;</div>
<div><span></span><br><br><tt><font size=3D"2">*allowing* (much less *requi=
ring*) the 199 to be reliable introduces<br>nasty problems. The 199 is only=
 an optimization, so having it be<br>unreliable is ok IMO.<br><br>The probl=
em is that if a proxy sends the 199, then the recipient of the<br>
PRACK should be the proxy. But the PRACK is an in-dialog message, so it<br>=
must be addressed to the Contact of the UAS.<br><br>If the proxy sends a re=
liable 199, and the PRACK is addressed to the<br>UAS, the UAS will be very =
surprised, since it has not send a reliable<br>
provisional, and is so not expecting a PRACK. In fact, it has sent a<br>fin=
al response, has presumably already received the ACK, and so is<br>expectin=
g nothing.<br><br>Very ugly.<br><br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; Paul<br><br><a onclick=3D"return top.js.OpenExtLink(window=
,event,this)" href=3D"mailto:wang.libo@zte.com.cn" target=3D"_blank">wang.l=
ibo@zte.com.cn</a> wrote:<br>
&gt; <br>&gt; Hi,<br>&gt; <br>&gt; And there still two questions left.<br>&=
gt; <br>&gt; 1. &nbsp;Is the 199 should be reliable or unreliable?<br>&gt; =
&nbsp; &nbsp; I think that 199 should be reliable if possible.<br>&gt; <br>=
&gt; <br>&gt; &nbsp; &nbsp;First, If a client receives an unreliable 199 re=
sponse on a dialog <br>
&gt; which has<br>&gt; &nbsp; &nbsp;not previously been created (this can h=
appen if a 199 response<br>&gt; &nbsp; &nbsp;reaches the client before a 18=
x response) the client SHALL discard<br>&gt; &nbsp; &nbsp;the 199 responses=
.<br>&gt; <br>&gt; &nbsp; &nbsp;The figure below shows an example.The 180 i=
s sent first but arrives <br>
&gt; later than 199.<br>&gt; If the 199 is reliable, the proxy should retra=
nsmit 199 (step 4),and <br>&gt; then the retransmitted<br>&gt; 199 will be =
accepted by UAC,and the early dialog will be teminated.<br>&gt; <br>&gt; UA=
C &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; P<br>
&gt; &nbsp; &nbsp;1. INVITE<br>&gt; ---------------&gt;<br>&gt; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; 2. 180<br>&gt; &lt;----- \/-------<br>&gt; &nbsp; &=
nbsp; &nbsp; &nbsp;/\ 3. 199<br>&gt; &lt;-----/ &nbsp;\------<br>&gt; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; 4. 199(retransmitted)<br>&gt; &lt;-----------=
----<br>
&gt; <br>&gt; &nbsp; &nbsp;second, According to practical use, 199 can be i=
ntended to teminate <br>&gt; one early dialog and release<br>&gt; resources=
 associated with the specific early dialog, so the cost spent <br>&gt; on r=
eliable 199 is worthy.<br>
&gt; If the 199 cannot be sent reliable,then we should send it unreliable.<=
br>&gt; &nbsp;<br>&gt; 2. In your last letter, you said &quot;the second 19=
9 could include <br>&gt; information which is to be forwarded to the<br>&gt=
; UAC&quot;, then do you mean, the early dialog is still alive after the fi=
rst <br>
&gt; 199 is accepted?<br>&gt; <br>&gt; <br>&gt; Also, In my last letter, I =
miss a &quot;NOT&quot; by mistake.<br>&gt; It should be<br>&gt; [Eric]: Sur=
ely the UAS is NOT allowed to send another reliable response<br>&gt; until =
the first one is acknowledged...<br>
&gt; <br>&gt; Regards,<br>&gt; Eric wang<br>&gt; <br>&gt; <br>&gt; <br>&gt;=
 <br>&gt; *&quot;Christer Holmberg&quot; &lt;<a onclick=3D"return top.js.Op=
enExtLink(window,event,this)" href=3D"mailto:christer.holmberg@ericsson.com=
" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;*<br>
&gt; <br>&gt; 2009-02-27 03:04<br>&gt; <br>&gt; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<br>&gt; =CA=D5=BC=FE=C8=CB<br>&gt; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&quot;Eric wang&q=
uot; &lt;<a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=
=3D"mailto:eric.wangxr@gmail.com" target=3D"_blank">eric.wangxr@gmail.com</=
a>&gt;<br>
&gt; =B3=AD=CB=CD<br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;&lt;<a onclick=3D"return top.js.OpenExtLink(window,event,this)=
" href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>&gt;, &lt;=
<a onclick=3D"return top.js.OpenExtLink(window,event,this)" href=3D"mailto:=
wang.libo@zte.com.cn" target=3D"_blank">wang.libo@zte.com.cn</a>&gt;<br>
&gt; =D6=F7=CC=E2<br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;RE: Questions about &quot;draft-ietf-sip-199-05&quot;<br>&gt; =
<br>&gt; <br>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;<br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; Hi,<br>&gt; <br>=
&gt; &nbsp;&gt;I still have some difficult in using IETF and cannot find th=
e whole<br>
&gt; sorted comments about this draft, so i have some doubt about this draf=
t.<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt; &nbsp; 1. &nbsp; &quot;If a forkin=
g proxy receives a reliably sent 199 response for a<br>&gt; dialog, for whi=
ch the proxy has previously generated and sent a 199<br>
&gt; response, the proxy MUST forward the 199 response.&quot;<br>&gt; &nbsp=
;&gt;<br>&gt; &nbsp;&gt; &nbsp; &nbsp; Does it describe the case below? Alt=
hough P1 have sent a 199<br>&gt; response, P1 havs to &nbsp;forword the sen=
d reliably 199 too.Or &nbsp;is the<br>
&gt; first 199 mistaked for 18x?<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp;UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; P1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;UA=
S_2<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;--- INVITE =
------&gt;<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; --- INVITE (leg 2) -&g=
t;<br>
&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- 199(leg 2) =
--<br>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;-- 199 (leg 2) -----<br>&g=
t; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- 199(leg 2) --=
<br>&gt; <br>&gt; &nbsp;&gt;I think it shall be 199, as currently written.<=
br>
&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;[Eric]: If &nbsp;this is 199, then Is the=
re a special purpose to send two<br>&gt; 199 in the same dialog?<br>&gt; &n=
bsp;&gt;I think that one 199 is enough.<br>&gt; &nbsp;&gt;And &nbsp;the fir=
st 199 may be reliable too, that will make it a little<br>
&gt; difficult to send the second 199.<br>&gt; <br>&gt; The second 199 coul=
d include information which is to be forwarded to the<br>&gt; UAC.<br>&gt; =
<br>&gt; <br>&gt; &nbsp;&gt;2. &nbsp;&quot;10. &nbsp;Usage with 100rel<br>&=
gt; &nbsp;&gt;<br>
&gt; &nbsp;&gt; &nbsp; When a 199 Early Dialog Terminated provisional respo=
nse is sent by a<br>&gt; UAS, since the provisional response is only used f=
or information<br>&gt; purpose, the UAS SHOULD send it unreliably even if t=
he 100rel<br>
&gt; &nbsp;&gt; &nbsp; option tag [RFC3262] is present in the Require heade=
r of the<br>&gt; associated request.&quot;<br>&gt; &nbsp;&gt;<br>&gt; &nbsp=
;&gt;<br>&gt; &nbsp;&gt;I have seen a comment on this question,but still no=
t understood about<br>
&gt; &nbsp;&gt;it. If the INVITE has a Require tag &quot;Require: 100rel&qu=
ot;,does the UAS<br>&gt; still<br>&gt; &nbsp;&gt;use unreliable 199 respons=
e?<br>&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;That is the recommendation, yes. Th=
e reasons is that we want to keep<br>
&gt; 199<br>&gt; &nbsp;&gt;as &quot;lightweight&quot; as possible, without =
requireing re-transmissions and<br>&gt; &nbsp;&gt;PRACKs.<br>&gt; &nbsp;&gt=
;<br>&gt; &nbsp;&gt;[Eric]: But reliable 199 have more advantage, and It is=
 worth to use<br>
&gt; the reliable 199,I think.<br>&gt; <br>&gt; I don&#39;t know what that =
advantage would be, compared to having to send<br>&gt; PRACKs etc. This has=
 been discussed quite much, so I would really need<br>&gt; some good justif=
ication to change it now.<br>
&gt; <br>&gt; Also, the draft doesn&#39;t forbid you to send the 199 reliab=
ly. It&#39;s only a<br>&gt; SHOULD.<br>&gt; <br>&gt; <br>&gt; &nbsp;&gt;If =
199 is reliable, there is one more advantage. If the 199 arrives<br>&gt; &n=
bsp;&gt;before the first 18x response, UAC can discard the first 199 and<br=
>
&gt; process<br>&gt; &nbsp;&gt;it until UAC receives the first 18x response=
 that has the same<br>&gt; &nbsp;&gt;to-tag as 199, as the reliable 199 sho=
uld be re-transmited until<br>&gt; &nbsp;&gt;received PRACK.<br>&gt; &nbsp;=
&gt;<br>&gt; &nbsp;&gt;If the INVITE contains &quot;Require: 100rel&quot;, =
the first 18x must also be<br>
&gt; &nbsp;&gt;sent reliably. And, I don&#39;t think the UAS is allowed to =
send another<br>&gt; &nbsp;&gt;reliable response until the first one is ack=
nowledged, so I don&#39;t think<br>&gt; &nbsp;&gt;a reliable 199 would reac=
h the UAC before the first reliable 18x.<br>
&gt; &nbsp;&gt;<br>&gt; &nbsp;&gt;[Eric]: Surely the UAS is allowed to send=
 another reliable response<br>&gt; until the first one is acknowledged,<br>=
&gt; <br>&gt; If I remember correctly, the FIRST reliable response must be =
acknowleded<br>
&gt; before the next reliable response is sent. But, we can double check in=
<br>&gt; the PRACK spec.<br>&gt; <br>&gt; &nbsp;&gt;but reliable 199 will b=
e useful if the first 18x is unreliable, or the<br>&gt; first 18x has been =
acknwledged. It&#39;s another case different from the<br>
&gt; above one &nbsp;whose INVITE contains &quot;require: 100rel&quot;.<br>=
&gt; <br>&gt; If 100rel is required the 18x cannot be unreliable.<br>&gt; <=
br>&gt; Regards,<br>&gt; <br>&gt; Christer<br>&gt; <br>&gt; <br>&gt; <br>&g=
t; <br>
&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; <br>&gt; ----------------=
----------------------------------------<br>&gt; ZTE Information Security N=
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</font></tt><br><br>&nbsp;</div><pre>--------------------------------------=
------------------
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From: "Eric.wang" <eric.wangxr@gmail.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>
References: <49A7484B.3060407@cisco.com> <OF331E0406.F68A068E-ON4825756A.001684FC-4825756A.00197C73@zte.com.cn> <CA9998CD4A020D418654FCDEF4E707DF0B167F1E@esealmw113.eemea.ericsson.se> <898632750902270249o3213670au6823cc4ab6180c8c@mail.gmail.com> <CA9998CD4A020D418654FCDEF4E707DF0B167F42@esealmw113.eemea.ericsson.se> <898632750902270310k72308b9au7645d26cfc2d7de8@mail.gmail.com> <CA9998CD4A020D418654FCDEF4E707DF0B167F46@esealmw113.eemea.ericsson.se>
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Subject: [Sip] =?utf-8?b?562U5aSNOiAg562U5aSNOiBSRTogUXVlc3Rpb25zIGFib3V0?= =?utf-8?q?_=22draft-ietf-sip-199-05=22?=
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Dear Christer,

=20

   It seems I=E2=80=99d better review all your comments firstJ.

 =20

   Thank you for answering my question.

=20

Best regards,

Eric

=20

=E5=8F=91=E4=BB=B6=E4=BA=BA: Christer Holmberg =
[mailto:christer.holmberg@ericsson.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2009=E5=B9=B42=E6=9C=8827=E6=97=A5 =
19:22
=E6=94=B6=E4=BB=B6=E4=BA=BA: Eric wang
=E6=8A=84=E9=80=81: wang.libo@zte.com.cn; Paul Kyzivat; sip@ietf.org
=E4=B8=BB=E9=A2=98: RE: [Sip] =E7=AD=94=E5=A4=8D: RE: Questions about =
"draft-ietf-sip-199-05"

=20

Hi,

>I don't think the 199 can be sent unreliably if UAC require 100rel.

>As in RFC3262, It says.

>  =20

>The UAS MUST send any non-100 provisional response reliably if the
>    initial request contained a Require header field with the option =
tag
>    100rel.  If the UAS is unwilling to do so, it MUST reject the =
initial
>    request with a 420 (Bad Extension) and include an Unsupported =
header
>    field containing the option tag 100rel.=20

=20

This has been discussed, and the proposed outcome was to allow 199 to be =
sent unreliably even if 100rel is required.

=20

Regards,

=20

Christer

=20

=20

=20

=20

=20

2009/2/27, Christer Holmberg <christer.holmberg@ericsson.com>:=20

Hi,

=20

>If the UAC requires 100rel ,then what should the proxy do if it =
receives a no-2xx   final response after forking?=20

=20

The proxy should send 199 unreliably.

=20

Regards,

=20

Christer=20

=20



=20

2009/2/27, Christer Holmberg <christer.holmberg@ericsson.com>:=20

Hi,

 =20
>Practically, when Call forward and fork are used in the same system, =
there are=20
>lots of early dialogs that should be eliminated by 199 existed.=20
>So I think that it's useful to let 199 reliable.=20
>And we just added fork-service to our system ,and find 199 is really =
useful, and=20
>if the 199 is missing, problems will happen. =20

=20

I'm glad you think 199 is useful :)

=20

But, if a proxy would have to terminate PRACKs etc it would not be a =
proxy anymore - it would be a B2BUA.

=20

So, if you really want your network to send 199 reliably, I guess you =
could use a B2BUA instead of a proxy.


>It sounds good that there is a way to prevent PRACK addressed to =
UAS,but i don't know  how to do now.=20

=20

The only way to prevent it is by the proxy not sending the 199 relaible. =
=20
   =20
>ps:=20
>RFC3261 16.7 Response Processing=20
>Since a proxy may not insert a tag into the To header field of=20
>a 1xx response to a request that did not contain one, it cannot=20
>issue non-100 provisional responses on its own. =20

=20

Yes, but I we have agreed that a proxy is allowed to send 199.

=20

But, the proxy is not going to generate new To header tag values for the =
199. It will use whatever tags that have already been created for the =
early dialogs.=20

=20

Regards,

=20

Christer


=20

=20

=20

=20

=20

=20

=20



*allowing* (much less *requiring*) the 199 to be reliable introduces
nasty problems. The 199 is only an optimization, so having it be
unreliable is ok IMO.

The problem is that if a proxy sends the 199, then the recipient of the
PRACK should be the proxy. But the PRACK is an in-dialog message, so it
must be addressed to the Contact of the UAS.

If the proxy sends a reliable 199, and the PRACK is addressed to the
UAS, the UAS will be very surprised, since it has not send a reliable
provisional, and is so not expecting a PRACK. In fact, it has sent a
final response, has presumably already received the ACK, and so is
expecting nothing.

Very ugly.

                Paul

wang.libo@zte.com.cn wrote:
>=20
> Hi,
>=20
> And there still two questions left.
>=20
> 1.  Is the 199 should be reliable or unreliable?
>     I think that 199 should be reliable if possible.
>=20
>=20
>    First, If a client receives an unreliable 199 response on a dialog=20
> which has
>    not previously been created (this can happen if a 199 response
>    reaches the client before a 18x response) the client SHALL discard
>    the 199 responses.
>=20
>    The figure below shows an example.The 180 is sent first but arrives =

> later than 199.
> If the 199 is reliable, the proxy should retransmit 199 (step 4),and=20
> then the retransmitted
> 199 will be accepted by UAC,and the early dialog will be teminated.
>=20
> UAC             P
>    1. INVITE
> --------------->
>           2. 180
> <----- \/-------
>        /\ 3. 199
> <-----/  \------
>           4. 199(retransmitted)
> <---------------
>=20
>    second, According to practical use, 199 can be intended to teminate =

> one early dialog and release
> resources associated with the specific early dialog, so the cost spent =

> on reliable 199 is worthy.
> If the 199 cannot be sent reliable,then we should send it unreliable.
> =20
> 2. In your last letter, you said "the second 199 could include=20
> information which is to be forwarded to the
> UAC", then do you mean, the early dialog is still alive after the =
first=20
> 199 is accepted?
>=20
>=20
> Also, In my last letter, I miss a "NOT" by mistake.
> It should be
> [Eric]: Surely the UAS is NOT allowed to send another reliable =
response
> until the first one is acknowledged...
>=20
> Regards,
> Eric wang
>=20
>=20
>=20
>=20
> *"Christer Holmberg" <christer.holmberg@ericsson.com>*
>=20
> 2009-02-27 03:04
>=20
>                 =20
> =E6=94=B6=E4=BB=B6=E4=BA=BA
>                  "Eric wang" <eric.wangxr@gmail.com>
> =E6=8A=84=E9=80=81
>                  <sip@ietf.org>, <wang.libo@zte.com.cn>
> =E4=B8=BB=E9=A2=98
>                  RE: Questions about "draft-ietf-sip-199-05"
>=20
>=20
>                 =20
>=20
>=20
>=20
>=20
>=20
> Hi,
>=20
>  >I still have some difficult in using IETF and cannot find the whole
> sorted comments about this draft, so i have some doubt about this =
draft.
>  >
>  >   1.   "If a forking proxy receives a reliably sent 199 response =
for a
> dialog, for which the proxy has previously generated and sent a 199
> response, the proxy MUST forward the 199 response."
>  >
>  >     Does it describe the case below? Although P1 have sent a 199
> response, P1 havs to  forword the send reliably 199 too.Or  is the
> first 199 mistaked for 18x?
>  >
>  >          UAC               P1                  UAS_2
>  >            --- INVITE ------>
>  >                             --- INVITE (leg 2) ->
>  >            <-- 199(leg 2) --
>  >                             <-- 199 (leg 2) -----
>  >            <-- 199(leg 2) --
>=20
>  >I think it shall be 199, as currently written.
>  >
>  >[Eric]: If  this is 199, then Is there a special purpose to send two
> 199 in the same dialog?
>  >I think that one 199 is enough.
>  >And  the first 199 may be reliable too, that will make it a little
> difficult to send the second 199.
>=20
> The second 199 could include information which is to be forwarded to =
the
> UAC.
>=20
>=20
>  >2.  "10.  Usage with 100rel
>  >
>  >   When a 199 Early Dialog Terminated provisional response is sent =
by a
> UAS, since the provisional response is only used for information
> purpose, the UAS SHOULD send it unreliably even if the 100rel
>  >   option tag [RFC3262] is present in the Require header of the
> associated request."
>  >
>  >
>  >I have seen a comment on this question,but still not understood =
about
>  >it. If the INVITE has a Require tag "Require: 100rel",does the UAS
> still
>  >use unreliable 199 response?
>  >
>  >That is the recommendation, yes. The reasons is that we want to keep
> 199
>  >as "lightweight" as possible, without requireing re-transmissions =
and
>  >PRACKs.
>  >
>  >[Eric]: But reliable 199 have more advantage, and It is worth to use
> the reliable 199,I think.
>=20
> I don't know what that advantage would be, compared to having to send
> PRACKs etc. This has been discussed quite much, so I would really need
> some good justification to change it now.
>=20
> Also, the draft doesn't forbid you to send the 199 reliably. It's only =
a
> SHOULD.
>=20
>=20
>  >If 199 is reliable, there is one more advantage. If the 199 arrives
>  >before the first 18x response, UAC can discard the first 199 and
> process
>  >it until UAC receives the first 18x response that has the same
>  >to-tag as 199, as the reliable 199 should be re-transmited until
>  >received PRACK.
>  >
>  >If the INVITE contains "Require: 100rel", the first 18x must also be
>  >sent reliably. And, I don't think the UAS is allowed to send another
>  >reliable response until the first one is acknowledged, so I don't =
think
>  >a reliable 199 would reach the UAC before the first reliable 18x.
>  >
>  >[Eric]: Surely the UAS is allowed to send another reliable response
> until the first one is acknowledged,
>=20
> If I remember correctly, the FIRST reliable response must be =
acknowleded
> before the next reliable response is sent. But, we can double check in
> the PRACK spec.
>=20
>  >but reliable 199 will be useful if the first 18x is unreliable, or =
the
> first 18x has been acknwledged. It's another case different from the
> above one  whose INVITE contains "require: 100rel".
>=20
> If 100rel is required the 18x cannot be unreliable.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> --------------------------------------------------------
> ZTE Information Security Notice: The information contained in this =
mail is solely property of the sender's organization. This mail =
communication is confidential. Recipients named above are obligated to =
maintain secrecy and are not permitted to disclose the contents of this =
communication to others.
> This email and any files transmitted with it are confidential and =
intended solely for the use of the individual or entity to whom they are =
addressed. If you have received this email in error please notify the =
originator of the message. Any views expressed in this message are those =
of the individual sender.
> This message has been scanned for viruses and Spam by ZTE Anti-Spam =
system.
>=20
>=20
> =
------------------------------------------------------------------------
>=20
> _______________________________________________
> Sip mailing list  https://www.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

--------------------------------------------------------
ZTE Information Security Notice: The information contained in this mail =
is solely property of the sender's organization. This mail communication =
is confidential. Recipients named above are obligated to maintain =
secrecy and are not permitted to disclose the contents of this =
communication to others.
This email and any files transmitted with it are confidential and =
intended solely for the use of the individual or entity to whom they are =
addressed. If you have received this email in error please notify the =
originator of the message. Any views expressed in this message are those =
of the individual sender.
This message has been scanned for viruses and Spam by ZTE Anti-Spam =
system.


_______________________________________________
Sip mailing list  https://www.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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<div class=3DSection1>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Dear Christer,<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>=C2=A0=C2=A0 It seems I=E2=80=99d better review all your =
comments first</span><span
lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:Wingdings;color:#1F497D'>J</span><s=
pan
lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>.<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>=C2=A0 <o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>=C2=A0=C2=A0 Thank you for answering my =
question.<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Best regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Eric<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

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

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt'>=E5=8F=91=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>:</span></span></b><span
lang=3DEN-US style=3D'font-size:10.0pt'> Christer Holmberg
[mailto:christer.holmberg@ericsson.com] <br>
</span><b><span =
style=3D'font-size:10.0pt'>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span =
lang=3DEN-US>:</span></span></b><span
lang=3DEN-US style=3D'font-size:10.0pt'> 2009</span><span =
style=3D'font-size:10.0pt'>=E5=B9=B4<span
lang=3DEN-US>2</span>=E6=9C=88<span lang=3DEN-US>27</span>=E6=97=A5<span =
lang=3DEN-US> 19:22<br>
</span><b>=E6=94=B6=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>:</span></b><span lang=3DEN-US> Eric wang<br>
</span><b>=E6=8A=84=E9=80=81<span lang=3DEN-US>:</span></b><span =
lang=3DEN-US>
wang.libo@zte.com.cn; Paul Kyzivat; sip@ietf.org<br>
</span><b>=E4=B8=BB=E9=A2=98<span lang=3DEN-US>:</span></b><span =
lang=3DEN-US> RE: [Sip] </span>=E7=AD=94=E5=A4=8D<span
lang=3DEN-US>: RE: Questions about =
&quot;draft-ietf-sip-199-05&quot;<o:p></o:p></span></span></p>

</div>

</div>

<p class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi,</span><span lang=3DEN-US><o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US>I don't think the 199 can be =
sent
unreliably if UAC require 100rel.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US>As&nbsp;in RFC3262, It =
says.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;&nbsp;</span><span =
lang=3DEN-US>&nbsp;&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US>The UAS MUST send any non-100
provisional response reliably if the<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US>&nbsp;&nbsp; initial request contained a Require header =
field with
the option tag<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US>&nbsp;&nbsp; 100rel.&nbsp; If the UAS is unwilling to do =
so, it MUST
reject the initial<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US>&nbsp;&nbsp; request with a 420 (Bad Extension) and include =
an
Unsupported header<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US>&nbsp;&nbsp; field containing the option tag =
100rel.</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>This&nbsp;has been&nbsp;discussed, and the proposed outcome =
was to
allow 199 to be sent unreliably even if 100rel is required.</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Christer</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span class=3Dgmailquote><span =
lang=3DEN-US>2009/2/27, Christer
Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt;:</span></span><span
lang=3DEN-US> <o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi,</span><span lang=3DEN-US><o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US>If the UAC requires 100rel =
,then what
should the proxy do if it receives a no-2xx</span><span lang=3DEN-US
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;
&nbsp;</span><span lang=3DEN-US>final response after =
forking?</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The proxy&nbsp;should send 199 unreliably.</span><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Christer</span><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US><br>
<br>
&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span class=3Dgmailquote><span =
lang=3DEN-US>2009/2/27, Christer
Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;:</span></span><s=
pan
lang=3DEN-US> <o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi,</span><span lang=3DEN-US><o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;&nbsp;<=
/span><span
lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>Practically, when Call forward and fork are used =
in the
same system, there are&nbsp;</span><span lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>lots of early dialogs that should be eliminated by =
199
existed.</span><span lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>So I think
that it's useful to let 199 reliable.&nbsp;</span><span =
lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>And we just added fork-service to our system ,and =
find
199 is really useful, and</span><span lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:black'>&gt;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>if the 199
is missing, problems will happen.</span><span =
lang=3DEN-US>&nbsp;</span><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>I'm glad you think 199 is useful :)</span><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>But, if a proxy would have to terminate PRACKs etc it would =
not be
a proxy anymore - it would be a B2BUA.</span><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>So, if you really want your network to send 199 reliably, I =
guess
you could use a B2BUA instead of a proxy.</span><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>It sounds good that there is a way to prevent =
PRACK
addressed to UAS,but i don't know<span style=3D'color:blue'>&nbsp; =
</span>how to
do now.<span style=3D'color:blue'>&nbsp;</span></span><span =
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The only way to prevent it is by the proxy not sending the
199&nbsp;relaible.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>&nbsp;</span><span lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;&nbsp;&=
nbsp;</span><span
lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US style=3D'font-family:"Times New =
Roman","serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>ps:</span><span lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US style=3D'font-family:"Times New =
Roman","serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>RFC3261 </span><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:"Courier New";color:#002041'>16.7 Response =
Processing</span></b><span
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=
<span
lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>Since a proxy may not insert a tag into the To =
header
field of</span><span lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif"'>a 1xx response to a request that did not contain =
one, <span
style=3D'color:blue'>it cannot</span></span><span =
lang=3DEN-US>&nbsp;<br>
</span><span lang=3DEN-US style=3D'font-family:"Times New =
Roman","serif";
color:black'>&gt;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Arial","sans-serif";color:blue'>issue non-100 provisional responses on =
its
own.</span><span lang=3DEN-US>&nbsp;</span><span lang=3DEN-US =
style=3D'font-size:
10.0pt;font-family:"Arial","sans-serif";color:blue'>&nbsp;</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Yes, but I&nbsp;we have agreed that a proxy is allowed to =
send 199.</span><span
lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>But, the proxy is not going to generate new To header tag =
values
for the 199. It will use whatever tags that have already&nbsp;been =
created for
the&nbsp;early dialogs.</span><span =
lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Christer</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US><br>
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;</span><span lang=3DEN-US><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span lang=3DEN-US><br>
<br>
</span><tt><span lang=3DEN-US style=3D'font-size:10.0pt'>*allowing* =
(much less
*requiring*) the 199 to be reliable introduces</span></tt><span =
lang=3DEN-US
style=3D'font-size:10.0pt'><br>
<tt>nasty problems. The 199 is only an optimization, so having it =
be</tt><br>
<tt>unreliable is ok IMO.</tt><br>
<br>
<tt>The problem is that if a proxy sends the 199, then the recipient of =
the</tt><br>
<tt>PRACK should be the proxy. But the PRACK is an in-dialog message, so =
it</tt><br>
<tt>must be addressed to the Contact of the UAS.</tt><br>
<br>
<tt>If the proxy sends a reliable 199, and the PRACK is addressed to =
the</tt><br>
<tt>UAS, the UAS will be very surprised, since it has not send a =
reliable</tt><br>
<tt>provisional, and is so not expecting a PRACK. In fact, it has sent =
a</tt><br>
<tt>final response, has presumably already received the ACK, and so =
is</tt><br>
<tt>expecting nothing.</tt><br>
<br>
<tt>Very ugly.</tt><br>
<br>
<tt>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Paul</tt><br>
<br>
<tt><a href=3D"mailto:wang.libo@zte.com.cn" =
target=3D"_blank">wang.libo@zte.com.cn</a>
wrote:</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Hi,</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; And there still two questions left.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; 1. &nbsp;Is the 199 should be reliable or unreliable?</tt><br>
<tt>&gt; &nbsp; &nbsp; I think that 199 should be reliable if =
possible.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp; &nbsp;First, If a client receives an unreliable 199 =
response on
a dialog </tt><br>
<tt>&gt; which has</tt><br>
<tt>&gt; &nbsp; &nbsp;not previously been created (this can happen if a =
199
response</tt><br>
<tt>&gt; &nbsp; &nbsp;reaches the client before a 18x response) the =
client
SHALL discard</tt><br>
<tt>&gt; &nbsp; &nbsp;the 199 responses.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp; &nbsp;The figure below shows an example.The 180 is sent =
first
but arrives </tt><br>
<tt>&gt; later than 199.</tt><br>
<tt>&gt; If the 199 is reliable, the proxy should retransmit 199 (step =
4),and </tt><br>
<tt>&gt; then the retransmitted</tt><br>
<tt>&gt; 199 will be accepted by UAC,and the early dialog will be =
teminated.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; P</tt><br>
<tt>&gt; &nbsp; &nbsp;1. INVITE</tt><br>
<tt>&gt; ---------------&gt;</tt><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 2. 180</tt><br>
<tt>&gt; &lt;----- \/-------</tt><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp;/\ 3. 199</tt><br>
<tt>&gt; &lt;-----/ &nbsp;\------</tt><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 4. =
199(retransmitted)</tt><br>
<tt>&gt; &lt;---------------</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp; &nbsp;second, According to practical use, 199 can be =
intended
to teminate </tt><br>
<tt>&gt; one early dialog and release</tt><br>
<tt>&gt; resources associated with the specific early dialog, so the =
cost spent
</tt><br>
<tt>&gt; on reliable 199 is worthy.</tt><br>
<tt>&gt; If the 199 cannot be sent reliable,then we should send it =
unreliable.</tt><br>
<tt>&gt; &nbsp;</tt><br>
<tt>&gt; 2. In your last letter, you said &quot;the second 199 could =
include </tt><br>
<tt>&gt; information which is to be forwarded to the</tt><br>
<tt>&gt; UAC&quot;, then do you mean, the early dialog is still alive =
after the
first </tt><br>
<tt>&gt; 199 is accepted?</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Also, In my last letter, I miss a &quot;NOT&quot; by =
mistake.</tt><br>
<tt>&gt; It should be</tt><br>
<tt>&gt; [Eric]: Surely the UAS is NOT allowed to send another reliable
response</tt><br>
<tt>&gt; until the first one is acknowledged...</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Regards,</tt><br>
<tt>&gt; Eric wang</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; *&quot;Christer Holmberg&quot; &lt;<a
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;*</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; 2009-02-27 03:04</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;</tt><br>
<tt>&gt; </tt></span><tt><span =
style=3D'font-size:10.0pt'>=E6=94=B6=E4=BB=B6=E4=BA=BA</span></tt><span
lang=3DEN-US style=3D'font-size:10.0pt'><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp;&quot;Eric wang&quot; &lt;<a href=3D"mailto:eric.wangxr@gmail.com"
target=3D"_blank">eric.wangxr@gmail.com</a>&gt;</tt><br>
<tt>&gt; </tt></span><tt><span =
style=3D'font-size:10.0pt'>=E6=8A=84=E9=80=81</span></tt><span
lang=3DEN-US style=3D'font-size:10.0pt'><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;&lt;<a
href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>&gt;, =
&lt;<a
href=3D"mailto:wang.libo@zte.com.cn" =
target=3D"_blank">wang.libo@zte.com.cn</a>&gt;</tt><br>
<tt>&gt; </tt></span><tt><span =
style=3D'font-size:10.0pt'>=E4=B8=BB=E9=A2=98</span></tt><span
lang=3DEN-US style=3D'font-size:10.0pt'><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;RE:
Questions about &quot;draft-ietf-sip-199-05&quot;</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Hi,</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp;&gt;I still have some difficult in using IETF and cannot =
find
the whole</tt><br>
<tt>&gt; sorted comments about this draft, so i have some doubt about =
this
draft.</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; 1. &nbsp; &quot;If a forking proxy receives a
reliably sent 199 response for a</tt><br>
<tt>&gt; dialog, for which the proxy has previously generated and sent a =
199</tt><br>
<tt>&gt; response, the proxy MUST forward the 199 =
response.&quot;</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; Does it describe the case below? =
Although P1
have sent a 199</tt><br>
<tt>&gt; response, P1 havs to &nbsp;forword the send reliably 199 too.Or
&nbsp;is the</tt><br>
<tt>&gt; first 199 mistaked for 18x?</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;UAC &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; P1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp;UAS_2</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;--- INVITE
------&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; --- INVITE (leg 2) =
-&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- =
199(leg 2)
--</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;-- 199 (leg 2) =
-----</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&lt;-- =
199(leg 2)
--</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp;&gt;I think it shall be 199, as currently =
written.</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;[Eric]: If &nbsp;this is 199, then Is there a special
purpose to send two</tt><br>
<tt>&gt; 199 in the same dialog?</tt><br>
<tt>&gt; &nbsp;&gt;I think that one 199 is enough.</tt><br>
<tt>&gt; &nbsp;&gt;And &nbsp;the first 199 may be reliable too, that =
will make
it a little</tt><br>
<tt>&gt; difficult to send the second 199.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; The second 199 could include information which is to be =
forwarded to
the</tt><br>
<tt>&gt; UAC.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp;&gt;2. &nbsp;&quot;10. &nbsp;Usage with 100rel</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; When a 199 Early Dialog Terminated =
provisional
response is sent by a</tt><br>
<tt>&gt; UAS, since the provisional response is only used for =
information</tt><br>
<tt>&gt; purpose, the UAS SHOULD send it unreliably even if the =
100rel</tt><br>
<tt>&gt; &nbsp;&gt; &nbsp; option tag [RFC3262] is present in the =
Require
header of the</tt><br>
<tt>&gt; associated request.&quot;</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;I have seen a comment on this question,but still not
understood about</tt><br>
<tt>&gt; &nbsp;&gt;it. If the INVITE has a Require tag &quot;Require:
100rel&quot;,does the UAS</tt><br>
<tt>&gt; still</tt><br>
<tt>&gt; &nbsp;&gt;use unreliable 199 response?</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;That is the recommendation, yes. The reasons is that =
we want
to keep</tt><br>
<tt>&gt; 199</tt><br>
<tt>&gt; &nbsp;&gt;as &quot;lightweight&quot; as possible, without =
requireing
re-transmissions and</tt><br>
<tt>&gt; &nbsp;&gt;PRACKs.</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;[Eric]: But reliable 199 have more advantage, and It =
is
worth to use</tt><br>
<tt>&gt; the reliable 199,I think.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; I don't know what that advantage would be, compared to having =
to send</tt><br>
<tt>&gt; PRACKs etc. This has been discussed quite much, so I would =
really need</tt><br>
<tt>&gt; some good justification to change it now.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Also, the draft doesn't forbid you to send the 199 reliably. =
It's only
a</tt><br>
<tt>&gt; SHOULD.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp;&gt;If 199 is reliable, there is one more advantage. If =
the 199
arrives</tt><br>
<tt>&gt; &nbsp;&gt;before the first 18x response, UAC can discard the =
first 199
and</tt><br>
<tt>&gt; process</tt><br>
<tt>&gt; &nbsp;&gt;it until UAC receives the first 18x response that has =
the
same</tt><br>
<tt>&gt; &nbsp;&gt;to-tag as 199, as the reliable 199 should be =
re-transmited
until</tt><br>
<tt>&gt; &nbsp;&gt;received PRACK.</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;If the INVITE contains &quot;Require: 100rel&quot;, =
the
first 18x must also be</tt><br>
<tt>&gt; &nbsp;&gt;sent reliably. And, I don't think the UAS is allowed =
to send
another</tt><br>
<tt>&gt; &nbsp;&gt;reliable response until the first one is =
acknowledged, so I
don't think</tt><br>
<tt>&gt; &nbsp;&gt;a reliable 199 would reach the UAC before the first =
reliable
18x.</tt><br>
<tt>&gt; &nbsp;&gt;</tt><br>
<tt>&gt; &nbsp;&gt;[Eric]: Surely the UAS is allowed to send another =
reliable
response</tt><br>
<tt>&gt; until the first one is acknowledged,</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; If I remember correctly, the FIRST reliable response must be
acknowleded</tt><br>
<tt>&gt; before the next reliable response is sent. But, we can double =
check in</tt><br>
<tt>&gt; the PRACK spec.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; &nbsp;&gt;but reliable 199 will be useful if the first 18x is
unreliable, or the</tt><br>
<tt>&gt; first 18x has been acknwledged. It's another case different =
from the</tt><br>
<tt>&gt; above one &nbsp;whose INVITE contains &quot;require: =
100rel&quot;.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; If 100rel is required the 18x cannot be unreliable.</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Regards,</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; Christer</tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
<tt>&gt; </tt><br>
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<p class=3DMsoNormal><span lang=3DEN-US><br>
_______________________________________________<br>
Sip mailing list &nbsp;<a =
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target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
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Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" =
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Use <a href=3D"mailto:sipping@ietf.org" =
target=3D"_blank">sipping@ietf.org</a> for
new developments on the application of sip<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span lang=3DEN-US><br>
<br>
<o:p></o:p></span></p>

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From: "Ian Elz" <ian.elz@ericsson.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <wang.libo@zte.com.cn>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Christer,

=20

The B2BUA should only send the 199 reliably if the Contact header has =
been mapped in the B2BUA.

=20

If the B2BUA has not mapped the Contact then you have a similar problem =
to that of the proxy.=20

=20

The B2BUA could of course intercept the PRACK and send the 200 OK even =
though the PRACK was addressed to the Contact provided in the initial =
response to the INVITE.

=20

To use Paul=A1=AFs words =A8C =A1=B0ugly=A1=B1.

=20

Ian Elz

________________________________

From: Christer Holmberg=20
Sent: 27 February 2009 06:23
To: wang.libo@zte.com.cn; Paul Kyzivat
Cc: sip@ietf.org
Subject: Re: [Sip]??: RE: Questions about "draft-ietf-sip-199-05"

=20

Hi,

 =20
>Practically, when Call forward and fork are used in the same system, =
there are=20
>lots of early dialogs that should be eliminated by 199 existed.=20
>So I think that it's useful to let 199 reliable.=20
>And we just added fork-service to our system ,and find 199 is really =
useful, and=20
>if the 199 is missing, problems will happen. =20

=20

I'm glad you think 199 is useful :)

=20

But, if a proxy would have to terminate PRACKs etc it would not be a =
proxy anymore - it would be a B2BUA.

=20

So, if you really want your network to send 199 reliably, I guess you =
could use a B2BUA instead of a proxy.


>It sounds good that there is a way to prevent PRACK addressed to =
UAS,but i don't know  how to do now.=20

=20

The only way to prevent it is by the proxy not sending the 199 relaible. =
=20
   =20
>ps:=20
>RFC3261 16.7 Response Processing=20
>Since a proxy may not insert a tag into the To header field of=20
>a 1xx response to a request that did not contain one, it cannot=20
>issue non-100 provisional responses on its own. =20

=20

Yes, but I we have agreed that a proxy is allowed to send 199.

=20

But, the proxy is not going to generate new To header tag values for the =
199. It will use whatever tags that have already been created for the =
early dialogs.=20

=20

Regards,

=20

Christer

=20
=20

=20

=20

=20

=20

=20

=20

=20

*allowing* (much less *requiring*) the 199 to be reliable introduces
nasty problems. The 199 is only an optimization, so having it be
unreliable is ok IMO.

The problem is that if a proxy sends the 199, then the recipient of the
PRACK should be the proxy. But the PRACK is an in-dialog message, so it
must be addressed to the Contact of the UAS.

If the proxy sends a reliable 199, and the PRACK is addressed to the
UAS, the UAS will be very surprised, since it has not send a reliable
provisional, and is so not expecting a PRACK. In fact, it has sent a
final response, has presumably already received the ACK, and so is
expecting nothing.

Very ugly.

                Paul

wang.libo@zte.com.cn wrote:
>=20
> Hi,
>=20
> And there still two questions left.
>=20
> 1.  Is the 199 should be reliable or unreliable?
>     I think that 199 should be reliable if possible.
>=20
>=20
>    First, If a client receives an unreliable 199 response on a dialog=20
> which has
>    not previously been created (this can happen if a 199 response
>    reaches the client before a 18x response) the client SHALL discard
>    the 199 responses.
>=20
>    The figure below shows an example.The 180 is sent first but arrives =

> later than 199.
> If the 199 is reliable, the proxy should retransmit 199 (step 4),and=20
> then the retransmitted
> 199 will be accepted by UAC,and the early dialog will be teminated.
>=20
> UAC             P
>    1. INVITE
> --------------->
>           2. 180
> <----- \/-------
>        /\ 3. 199
> <-----/  \------
>           4. 199(retransmitted)
> <---------------
>=20
>    second, According to practical use, 199 can be intended to teminate =

> one early dialog and release
> resources associated with the specific early dialog, so the cost spent =

> on reliable 199 is worthy.
> If the 199 cannot be sent reliable,then we should send it unreliable.
> =20
> 2. In your last letter, you said "the second 199 could include=20
> information which is to be forwarded to the
> UAC", then do you mean, the early dialog is still alive after the =
first=20
> 199 is accepted?
>=20
>=20
> Also, In my last letter, I miss a "NOT" by mistake.
> It should be
> [Eric]: Surely the UAS is NOT allowed to send another reliable =
response
> until the first one is acknowledged...
>=20
> Regards,
> Eric wang
>=20
>=20
>=20
>=20
> *"Christer Holmberg" <christer.holmberg@ericsson.com>*
>=20
> 2009-02-27 03:04
>=20
>                 =20
> =CA=D5=BC=FE=C8=CB
>                  "Eric wang" <eric.wangxr@gmail.com>
> =B3=AD=CB=CD
>                  <sip@ietf.org>, <wang.libo@zte.com.cn>
> =D6=F7=CC=E2
>                  RE: Questions about "draft-ietf-sip-199-05"
>=20
>=20
>                 =20
>=20
>=20
>=20
>=20
>=20
> Hi,
>=20
>  >I still have some difficult in using IETF and cannot find the whole
> sorted comments about this draft, so i have some doubt about this =
draft.
>  >
>  >   1.   "If a forking proxy receives a reliably sent 199 response =
for a
> dialog, for which the proxy has previously generated and sent a 199
> response, the proxy MUST forward the 199 response."
>  >
>  >     Does it describe the case below? Although P1 have sent a 199
> response, P1 havs to  forword the send reliably 199 too.Or  is the
> first 199 mistaked for 18x?
>  >
>  >          UAC               P1                  UAS_2
>  >            --- INVITE ------>
>  >                             --- INVITE (leg 2) ->
>  >            <-- 199(leg 2) --
>  >                             <-- 199 (leg 2) -----
>  >            <-- 199(leg 2) --
>=20
>  >I think it shall be 199, as currently written.
>  >
>  >[Eric]: If  this is 199, then Is there a special purpose to send two
> 199 in the same dialog?
>  >I think that one 199 is enough.
>  >And  the first 199 may be reliable too, that will make it a little
> difficult to send the second 199.
>=20
> The second 199 could include information which is to be forwarded to =
the
> UAC.
>=20
>=20
>  >2.  "10.  Usage with 100rel
>  >
>  >   When a 199 Early Dialog Terminated provisional response is sent =
by a
> UAS, since the provisional response is only used for information
> purpose, the UAS SHOULD send it unreliably even if the 100rel
>  >   option tag [RFC3262] is present in the Require header of the
> associated request."
>  >
>  >
>  >I have seen a comment on this question,but still not understood =
about
>  >it. If the INVITE has a Require tag "Require: 100rel",does the UAS
> still
>  >use unreliable 199 response?
>  >
>  >That is the recommendation, yes. The reasons is that we want to keep
> 199
>  >as "lightweight" as possible, without requireing re-transmissions =
and
>  >PRACKs.
>  >
>  >[Eric]: But reliable 199 have more advantage, and It is worth to use
> the reliable 199,I think.
>=20
> I don't know what that advantage would be, compared to having to send
> PRACKs etc. This has been discussed quite much, so I would really need
> some good justification to change it now.
>=20
> Also, the draft doesn't forbid you to send the 199 reliably. It's only =
a
> SHOULD.
>=20
>=20
>  >If 199 is reliable, there is one more advantage. If the 199 arrives
>  >before the first 18x response, UAC can discard the first 199 and
> process
>  >it until UAC receives the first 18x response that has the same
>  >to-tag as 199, as the reliable 199 should be re-transmited until
>  >received PRACK.
>  >
>  >If the INVITE contains "Require: 100rel", the first 18x must also be
>  >sent reliably. And, I don't think the UAS is allowed to send another
>  >reliable response until the first one is acknowledged, so I don't =
think
>  >a reliable 199 would reach the UAC before the first reliable 18x.
>  >
>  >[Eric]: Surely the UAS is allowed to send another reliable response
> until the first one is acknowledged,
>=20
> If I remember correctly, the FIRST reliable response must be =
acknowleded
> before the next reliable response is sent. But, we can double check in
> the PRACK spec.
>=20
>  >but reliable 199 will be useful if the first 18x is unreliable, or =
the
> first 18x has been acknwledged. It's another case different from the
> above one  whose INVITE contains "require: 100rel".
>=20
> If 100rel is required the 18x cannot be unreliable.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> --------------------------------------------------------
> ZTE Information Security Notice: The information contained in this =
mail is solely property of the sender's organization. This mail =
communication is confidential. Recipients named above are obligated to =
maintain secrecy and are not permitted to disclose the contents of this =
communication to others.
> This email and any files transmitted with it are confidential and =
intended solely for the use of the individual or entity to whom they are =
addressed. If you have received this email in error please notify the =
originator of the message. Any views expressed in this message are those =
of the individual sender.
> This message has been scanned for viruses and Spam by ZTE Anti-Spam =
system.
>=20
>=20
> =
------------------------------------------------------------------------
>=20
> _______________________________________________
> Sip mailing list  https://www.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



--------------------------------------------------------
ZTE Information Security Notice: The information contained in this mail =
is solely property of the sender's organization. This mail communication =
is confidential. Recipients named above are obligated to maintain =
secrecy and are not permitted to disclose the contents of this =
communication to others.
This email and any files transmitted with it are confidential and =
intended solely for the use of the individual or entity to whom they are =
addressed. If you have received this email in error please notify the =
originator of the message. Any views expressed in this message are those =
of the individual sender.
This message has been scanned for viruses and Spam by ZTE Anti-Spam =
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</head>

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

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Christer,<o:p></o:p></span></font></=
p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The B2BUA should only send the 199
reliably if the Contact header has been mapped in the =
B2BUA.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If the B2BUA has not mapped the =
Contact
then you have a similar problem to that of the proxy. =
<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The B2BUA could of course intercept =
the
PRACK and send the 200 OK even though the PRACK was addressed to the =
Contact
provided in the initial response to the =
INVITE.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>To use Paul=A1=AFs words =A8C =
=A1=B0ugly=A1=B1.<o:p></o:p></span></font></p>

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

<div>

<p><st1:PersonName w:st=3D"on"><i><font size=3D2 color=3Dnavy =
face=3DArial><span
 lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial;color:navy;font-style:
 italic'>Ian Elz</span></font></i></st1:PersonName><font =
color=3Dnavy><span
style=3D'color:navy'><o:p></o:p></span></font></p>

</div>

<div>

<div class=3DMsoNormal align=3Dcenter =
style=3D'margin-left:36.0pt;text-align:center'><font
size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> <st1:PersonName
w:st=3D"on">Christer Holmberg</st1:PersonName> <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 27 February 2009 =
06:23<br>
<b><span style=3D'font-weight:bold'>To:</span></b> wang.libo@zte.com.cn; =
Paul
Kyzivat<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sip]??: RE:
Questions about =
&quot;draft-ietf-sip-199-05&quot;</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Hi,</span></font>=
<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
face=3Dsans-serif><span
style=3D'font-size:10.0pt;font-family:sans-serif'>&nbsp;&nbsp;</span></fo=
nt><br>
<font size=3D2 color=3Dblack face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:black'>&gt;</span></font><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Practically, when Call =
forward and
fork are used in the same system, there are&nbsp;</span></font><br>
<font size=3D2 color=3Dblack face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:black'>&gt;</span></font><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>lots of early dialogs that =
should be
eliminated by 199 existed.</span></font>&nbsp;<br>
<font color=3Dblack face=3DArial><span =
style=3D'font-family:Arial;color:black'>&gt;</span></font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>So I think
that it's useful to let 199 reliable.&nbsp;</span></font><br>
<font size=3D2 color=3Dblack face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:black'>&gt;</span></font><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>And we just added =
fork-service to
our system ,and find 199 is really useful, and</span></font>&nbsp;<br>
<font color=3Dblack face=3DArial><span =
style=3D'font-family:Arial;color:black'>&gt;</span></font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>if the 199
is missing, problems will happen.</span></font>&nbsp;<font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>&nbsp;</span></fo=
nt><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>I'm glad
you think 199 is useful :)</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>But, if
a proxy would have to terminate PRACKs etc it would not be a proxy =
anymore - it
would be a B2BUA.</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>So, if
you really want your network to send 199 reliably, I guess you could use =
a
B2BUA instead of a proxy.</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
</span></font><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:blue'>&gt;</span></font><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>It sounds good that there =
is a way
to prevent PRACK addressed to UAS,but i don't know<font =
color=3Dblue><span
style=3D'color:blue'>&nbsp; </span></font>how to do now.<font =
color=3Dblue><span
style=3D'color:blue'>&nbsp;</span></font></span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>The only
way to prevent it is by the proxy not sending the =
199&nbsp;relaible.</span></font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>&nbsp;<b=
r>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>&nbsp;&nbsp;&nbsp;</spa=
n></font>&nbsp;<br>
<font color=3Dblack><span style=3D'color:black'>&gt;</span></font><font =
size=3D2
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>ps:</span></font>&nbsp;<br>
<font color=3Dblack><span style=3D'color:black'>&gt;</span></font><font =
size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>RFC3261 =
</span></font><b><font
size=3D2 color=3D"#002041" face=3D"Courier New"><span =
style=3D'font-size:10.0pt;
font-family:"Courier New";color:#002041;font-weight:bold'>16.7 Response
Processing</span></font></b><font size=3D2 face=3Dsans-serif><span
style=3D'font-size:10.0pt;font-family:sans-serif'>&nbsp;</span></font><br=
>
<font size=3D2 color=3Dblack face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:black'>&gt;</span></font><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Since a proxy may not =
insert a tag
into the To header field of</span></font>&nbsp;<br>
<font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:blue'>&gt;</span></font><font size=3D2 face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>a 1xx response to a request =
that did
not contain one, <font color=3Dblue><span style=3D'color:blue'>it =
cannot</span></font></span></font>&nbsp;<br>
<font color=3Dblack><span style=3D'color:black'>&gt;</span></font><font =
size=3D2
color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:blue'>issue non-100 provisional responses on its =
own.</span></font>&nbsp;<font
size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:blue'>&nbsp;</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Yes, but
I&nbsp;we have agreed that a proxy is allowed to send =
199.</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>But, the
proxy is not going to generate new To header tag values for the 199. It =
will
use whatever tags that have already&nbsp;been created for the&nbsp;early
dialogs.</span></font>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>&nbsp;</span></fo=
nt><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Regards,</span></=
font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue'>Christer</span></=
font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;<br>
</span></font><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:blue'>&nbsp;</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:36.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:
12.0pt;margin-left:36.0pt'><font size=3D3 face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;<br>
<br>
</span></font><tt><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>*allowing*
(much less *requiring*) the 199 to be reliable =
introduces</span></font></tt><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">nasty problems. The 199 is only an =
optimization,
so having it be</font></tt><br>
<tt><font face=3D"Courier New">unreliable is ok IMO.</font></tt><br>
<br>
<tt><font face=3D"Courier New">The problem is that if a proxy sends the =
199, then
the recipient of the</font></tt><br>
<tt><font face=3D"Courier New">PRACK should be the proxy. But the PRACK =
is an
in-dialog message, so it</font></tt><br>
<tt><font face=3D"Courier New">must be addressed to the Contact of the =
UAS.</font></tt><br>
<br>
<tt><font face=3D"Courier New">If the proxy sends a reliable 199, and =
the PRACK
is addressed to the</font></tt><br>
<tt><font face=3D"Courier New">UAS, the UAS will be very surprised, =
since it has
not send a reliable</font></tt><br>
<tt><font face=3D"Courier New">provisional, and is so not expecting a =
PRACK. In
fact, it has sent a</font></tt><br>
<tt><font face=3D"Courier New">final response, has presumably already =
received
the ACK, and so is</font></tt><br>
<tt><font face=3D"Courier New">expecting nothing.</font></tt><br>
<br>
<tt><font face=3D"Courier New">Very ugly.</font></tt><br>
<br>
<tt><font face=3D"Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; Paul</font></tt><br>
<br>
<tt><font face=3D"Courier New">wang.libo@zte.com.cn =
wrote:</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Hi,</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; And there still two questions =
left.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; 1. &nbsp;Is the 199 should be =
reliable or
unreliable?</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; I think that 199 =
should be
reliable if possible.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;First, If a client =
receives an
unreliable 199 response on a dialog </font></tt><br>
<tt><font face=3D"Courier New">&gt; which has</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;not previously been =
created
(this can happen if a 199 response</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;reaches the client =
before a 18x
response) the client SHALL discard</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;the 199 =
responses.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;The figure below shows =
an
example.The 180 is sent first but arrives </font></tt><br>
<tt><font face=3D"Courier New">&gt; later than 199.</font></tt><br>
<tt><font face=3D"Courier New">&gt; If the 199 is reliable, the proxy =
should
retransmit 199 (step 4),and </font></tt><br>
<tt><font face=3D"Courier New">&gt; then the =
retransmitted</font></tt><br>
<tt><font face=3D"Courier New">&gt; 199 will be accepted by UAC,and the =
early
dialog will be teminated.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; UAC &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
P</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;1. =
INVITE</font></tt><br>
<tt><font face=3D"Courier New">&gt; ---------------&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
2. 180</font></tt><br>
<tt><font face=3D"Courier New">&gt; &lt;----- \/-------</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp;/\ 3. =
199</font></tt><br>
<tt><font face=3D"Courier New">&gt; &lt;-----/ =
&nbsp;\------</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
4. 199(retransmitted)</font></tt><br>
<tt><font face=3D"Courier New">&gt; &lt;---------------</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp;second, According to =
practical
use, 199 can be intended to teminate </font></tt><br>
<tt><font face=3D"Courier New">&gt; one early dialog and =
release</font></tt><br>
<tt><font face=3D"Courier New">&gt; resources associated with the =
specific early
dialog, so the cost spent </font></tt><br>
<tt><font face=3D"Courier New">&gt; on reliable 199 is =
worthy.</font></tt><br>
<tt><font face=3D"Courier New">&gt; If the 199 cannot be sent =
reliable,then we
should send it unreliable.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;</font></tt><br>
<tt><font face=3D"Courier New">&gt; 2. In your last letter, you said =
&quot;the
second 199 could include </font></tt><br>
<tt><font face=3D"Courier New">&gt; information which is to be forwarded =
to the</font></tt><br>
<tt><font face=3D"Courier New">&gt; UAC&quot;, then do you mean, the =
early dialog
is still alive after the first </font></tt><br>
<tt><font face=3D"Courier New">&gt; 199 is accepted?</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Also, In my last letter, I miss a
&quot;NOT&quot; by mistake.</font></tt><br>
<tt><font face=3D"Courier New">&gt; It should be</font></tt><br>
<tt><font face=3D"Courier New">&gt; [Eric]: Surely the UAS is NOT =
allowed to send
another reliable response</font></tt><br>
<tt><font face=3D"Courier New">&gt; until the first one is =
acknowledged...</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Regards,</font></tt><br>
<tt><font face=3D"Courier New">&gt; Eric wang</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; *&quot;<st1:PersonName =
w:st=3D"on">Christer
 Holmberg</st1:PersonName>&quot; =
&lt;christer.holmberg@ericsson.com&gt;*</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; 2009-02-27 03:04</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp;</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt></span></font><tt><font =
size=3D2
face=3D"MS Mincho"><span style=3D'font-size:10.0pt;font-family:"MS =
Mincho"'>=CA=D5=BC=FE=C8=CB</span></font></tt><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp;&quot;Eric wang&quot; =
&lt;eric.wangxr@gmail.com&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt></span></font><tt><font =
size=3D2
face=3D"MS Mincho"><span style=3D'font-size:10.0pt;font-family:"MS =
Mincho"'>=B3=AD=CB=CD</span></font></tt><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp;&lt;sip@ietf.org&gt;, =
&lt;wang.libo@zte.com.cn&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt></span></font><tt><font =
size=3D2
face=3D"MS Mincho"><span style=3D'font-size:10.0pt;font-family:"MS =
Mincho"'>=D6=F7</span></font></tt><tt><font
size=3D2 face=3DSimSun><span =
style=3D'font-size:10.0pt;font-family:SimSun'>=CC=E2</span></font></tt><f=
ont
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp;RE: Questions about =
&quot;draft-ietf-sip-199-05&quot;</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp;</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Hi,</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;I still have some =
difficult in
using IETF and cannot find the whole</font></tt><br>
<tt><font face=3D"Courier New">&gt; sorted comments about this draft, so =
i have
some doubt about this draft.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; 1. &nbsp; &quot;If =
a forking
proxy receives a reliably sent 199 response for a</font></tt><br>
<tt><font face=3D"Courier New">&gt; dialog, for which the proxy has =
previously
generated and sent a 199</font></tt><br>
<tt><font face=3D"Courier New">&gt; response, the proxy MUST forward the =
199
response.&quot;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; Does it =
describe the
case below? Although P1 have sent a 199</font></tt><br>
<tt><font face=3D"Courier New">&gt; response, P1 havs to &nbsp;forword =
the send
reliably 199 too.Or &nbsp;is the</font></tt><br>
<tt><font face=3D"Courier New">&gt; first 199 mistaked for =
18x?</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;
&nbsp;UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; P1 &nbsp; =
&nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;UAS_2</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
&nbsp;--- INVITE ------&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; --- =
INVITE (leg
2) -&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
&nbsp;&lt;-- 199(leg 2) --</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &lt;-- =
199 (leg
2) -----</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;
&nbsp;&lt;-- 199(leg 2) --</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;I think it shall be 199, =
as
currently written.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;[Eric]: If &nbsp;this is =
199, then
Is there a special purpose to send two</font></tt><br>
<tt><font face=3D"Courier New">&gt; 199 in the same =
dialog?</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;I think that one 199 is =
enough.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;And &nbsp;the first 199 =
may be
reliable too, that will make it a little</font></tt><br>
<tt><font face=3D"Courier New">&gt; difficult to send the second =
199.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; The second 199 could include =
information
which is to be forwarded to the</font></tt><br>
<tt><font face=3D"Courier New">&gt; UAC.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;2. &nbsp;&quot;10. =
&nbsp;Usage with
100rel</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; When a 199 Early =
Dialog
Terminated provisional response is sent by a</font></tt><br>
<tt><font face=3D"Courier New">&gt; UAS, since the provisional response =
is only
used for information</font></tt><br>
<tt><font face=3D"Courier New">&gt; purpose, the UAS SHOULD send it =
unreliably
even if the 100rel</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt; &nbsp; option tag =
[RFC3262] is
present in the Require header of the</font></tt><br>
<tt><font face=3D"Courier New">&gt; associated =
request.&quot;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;I have seen a comment on =
this
question,but still not understood about</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;it. If the INVITE has a =
Require tag
&quot;Require: 100rel&quot;,does the UAS</font></tt><br>
<tt><font face=3D"Courier New">&gt; still</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;use unreliable 199 =
response?</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;That is the =
recommendation, yes.
The reasons is that we want to keep</font></tt><br>
<tt><font face=3D"Courier New">&gt; 199</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;as &quot;lightweight&quot; =
as
possible, without requireing re-transmissions and</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;PRACKs.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;[Eric]: But reliable 199 =
have more
advantage, and It is worth to use</font></tt><br>
<tt><font face=3D"Courier New">&gt; the reliable 199,I =
think.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; I don't know what that advantage =
would be,
compared to having to send</font></tt><br>
<tt><font face=3D"Courier New">&gt; PRACKs etc. This has been discussed =
quite
much, so I would really need</font></tt><br>
<tt><font face=3D"Courier New">&gt; some good justification to change it =
now.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Also, the draft doesn't forbid you =
to send
the 199 reliably. It's only a</font></tt><br>
<tt><font face=3D"Courier New">&gt; SHOULD.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;If 199 is reliable, there =
is one
more advantage. If the 199 arrives</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;before the first 18x =
response, UAC
can discard the first 199 and</font></tt><br>
<tt><font face=3D"Courier New">&gt; process</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;it until UAC receives the =
first 18x
response that has the same</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;to-tag as 199, as the =
reliable 199
should be re-transmited until</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;received =
PRACK.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;If the INVITE contains
&quot;Require: 100rel&quot;, the first 18x must also be</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;sent reliably. And, I =
don't think
the UAS is allowed to send another</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;reliable response until =
the first
one is acknowledged, so I don't think</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;a reliable 199 would reach =
the UAC
before the first reliable 18x.</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;</font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;[Eric]: Surely the UAS is =
allowed
to send another reliable response</font></tt><br>
<tt><font face=3D"Courier New">&gt; until the first one is =
acknowledged,</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; If I remember correctly, the FIRST =
reliable
response must be acknowleded</font></tt><br>
<tt><font face=3D"Courier New">&gt; before the next reliable response is =
sent.
But, we can double check in</font></tt><br>
<tt><font face=3D"Courier New">&gt; the PRACK spec.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; &nbsp;&gt;but reliable 199 will be =
useful if
the first 18x is unreliable, or the</font></tt><br>
<tt><font face=3D"Courier New">&gt; first 18x has been acknwledged. It's =
another
case different from the</font></tt><br>
<tt><font face=3D"Courier New">&gt; above one &nbsp;whose INVITE =
contains
&quot;require: 100rel&quot;.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; If 100rel is required the 18x cannot =
be
unreliable.</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Regards,</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; Christer</font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt; </font></tt><br>
<tt><font face=3D"Courier New">&gt;
--------------------------------------------------------</font></tt><br>
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_______________________________________________</font></tt><br>
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</span></font><o:p></o:p></p>

</div>

<pre style=3D'margin-left:36.0pt'><font size=3D2 face=3D"Courier =
New"><span
style=3D'font-size:10.0pt'>----------------------------------------------=
----------<o:p></o:p></span></font></pre><pre
style=3D'margin-left:36.0pt'><font size=3D2 face=3D"Courier New"><span
style=3D'font-size:10.0pt'>ZTE&nbsp;Information&nbsp;Security&nbsp;Notice=
:&nbsp;The&nbsp;information&nbsp;contained&nbsp;in&nbsp;this&nbsp;mail&nb=
sp;is&nbsp;solely&nbsp;property&nbsp;of&nbsp;the&nbsp;sender's&nbsp;organ=
ization.&nbsp;This&nbsp;mail&nbsp;communication&nbsp;is&nbsp;confidential=
.&nbsp;Recipients&nbsp;named&nbsp;above&nbsp;are&nbsp;obligated&nbsp;to&n=
bsp;maintain&nbsp;secrecy&nbsp;and&nbsp;are&nbsp;not&nbsp;permitted&nbsp;=
to&nbsp;disclose&nbsp;the&nbsp;contents&nbsp;of&nbsp;this&nbsp;communicat=
ion&nbsp;to&nbsp;others.<o:p></o:p></span></font></pre><pre
style=3D'margin-left:36.0pt'><font size=3D2 face=3D"Courier New"><span
style=3D'font-size:10.0pt'>This&nbsp;email&nbsp;and&nbsp;any&nbsp;files&n=
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From alan@sipstation.com  Fri Feb 27 08:46:53 2009
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Date: Fri, 27 Feb 2009 10:47:12 -0600
From: Alan Johnston <alan@sipstation.com>
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Subject: Re: [Sip] Questions about the status of dual video feature in SIP
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Hi Yang,

Based on a little reading on H.239, it sounds like it is a conference 
feature set, which would be implemented using SIP, RTP, various SDP 
extensions for media labeling and grouping, and XCON protocols such as 
BFCP for floor control and possibly CCMP for conference control.

As such, you won't find one document that describes all these protocols, 
and most of the action happens outside of SIP.  If you have trouble 
finding some of these protocols, let me know.

Also, discussion on this would probably be most productive on the XCON list.

Thanks,
Alan


poplarbird wrote:
>
> Hi All,
>
>  
>
> I am currently working on the interworking between SIP and H323.
>
>  
>
> I had difficult in finding the status of SIP supporting dual video 
> (e.g. H.239 from a H.323 endpoint).
>
> Are there any drafts or standards working on it?
>
>  
>
> Also with the text chatting in a SIP call, if anyone can give me the 
> RFC number that would be great.
>
>  
>
> Thanks,
>
> Yang
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Sip mailing list  https://www.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 christer.holmberg@ericsson.com  Fri Feb 27 11:53:29 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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X-OriginalArrivalTime: 27 Feb 2009 19:51:10.0485 (UTC) FILETIME=[BCED8C50:01C99914]
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Framework for Establishing an SRTP Security Context using DTLS
	Author(s)       : J. Fischl, et al.
	Filename        : draft-ietf-sip-dtls-srtp-framework-06.txt
	Pages           : 38
	Date            : 2009-02-28

This document specifies how to use the Session Initiation Protocol
(SIP) to establish an Secure Real-time Transport Protocol (SRTP)
security context using the Datagram Transport Layer Security (DTLS)
protocol.  It describes a mechanism of transporting a fingerprint
attribute in the Session Description Protocol (SDP) that identifies
the key that will be presented during the DTLS handshake.  The key
exchange travels along the media path as opposed to the signaling
path.  The SIP Identity mechanism can be used to protect the
integrity of the fingerprint attribute from modification by
intermediate proxies.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-dtls-srtp-framework-06.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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