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From: Lakshmi <lakshmi@multitech.co.in>
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Subject: [Sip] Need clarification on video with sip stack
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Hello,

I am Lakshmi and a silent member of this mailing group. I need some 
clarification on sip stack. I am beginer of sip and got some work to 
work with sip.
The present sip stack what we have is only carries voice information. I 
have to enhance this voice phone to video phone. Means needs to 
implement video with sip stack so that sip stack should support this 
video featuer with voice phone.

According to my analysis , SIP doesn't require any modifications since 
sip concentrates on the session part and doesn't bother about what kind 
of session it was.
And sdp needs to change to do video parameters negotiation (Hope RTP 
payload also). Kindly clarify this concept or guide me how to proceed to 
accomplish this task. And also please correct if my analysis is wrong.

Regards,
-Lakshmi


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



From sip-bounces@ietf.org Mon Jul 02 02:24:00 2007
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Subject: RE: [Sip] Need clarification on video with sip stack
Date: Mon, 2 Jul 2007 11:54:28 +0530
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From: <kamalakar.mergu@wipro.com>
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Hi Lakshmi,
All the changes required in SIP stack is w.r.t addition of SDP
capabilities to support and negotiate Video codecs and related stuff.
SIP stack does not bother about RTP payload at all.
It is upto your video application to code and decode the rtp payload .

Regards
Kamal


-----Original Message-----
From: Lakshmi [mailto:lakshmi@multitech.co.in]=20
Sent: Sunday, July 01, 2007 9:33 AM
To: sip@ietf.org
Subject: [Sip] Need clarification on video with sip stack

Hello,

I am Lakshmi and a silent member of this mailing group. I need some=20
clarification on sip stack. I am beginer of sip and got some work to=20
work with sip.
The present sip stack what we have is only carries voice information. I=20
have to enhance this voice phone to video phone. Means needs to=20
implement video with sip stack so that sip stack should support this=20
video featuer with voice phone.

According to my analysis , SIP doesn't require any modifications since=20
sip concentrates on the session part and doesn't bother about what kind=20
of session it was.
And sdp needs to change to do video parameters negotiation (Hope RTP=20
payload also). Kindly clarify this concept or guide me how to proceed to

accomplish this task. And also please correct if my analysis is wrong.

Regards,
-Lakshmi


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


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



From sip-bounces@ietf.org Mon Jul 02 03:33:25 2007
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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I think we should put a couple of "do we charter" items on the agenda:
=20
These are the mime body handling draft from Gonzalo, and the proposed
INFO handling draft from Eric.
=20
So far these are the only two that I can identify as having had
discussion and for which we do not have slots.
=20
Not sure whether we need a slot for corrections or not - I think Robert
was suggesting an ad hoc somewhere to progress this.
=20
regards
=20
Keith


________________________________

	From: Dean Willis [mailto:dean.willis@softarmor.com]=20
	Sent: Sunday, July 01, 2007 2:51 AM
	To: IETF SIP
	Cc: Cullen Jennings; DRAGE, Keith (Keith)
	Subject: Draft Agenda for SIP at IETF 69 in Chicago
=09
=09
	Here's the current draft agenda for SIP at IETF 69. I'm sure it
is incomplete. Changes will be posted to:
	http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69


	Draft SIP Agenda IETF 69

=09
=09

	Session 1, Monday July 23 0900-1130 Grand Ballroom

Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0915
SIPS WGLC
Francois Audet
draft-ietf-sip-sips-05.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt>=20
0945
Outbound
Rohan Mahy
Cullen Jennings
draft-ietf-sip-outbound-09.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt>=20
1015
Location Conveyance WGLC
James Polk
draft-ietf-sip-location-conveyance-07.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-
07.txt>=20
1045
Delivering R-URI and Parameters to UA
Jonathan Rosenberg
draft-rosenberg-sip-ua-loose-route-01.txt
<http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-route-
01.txt>=20
1130
End of Session





=09
=09

	Session 2, Tuesday July 24 0900-1130 Grand Ballroom

Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0905
SAML
Hannes Tschofenig
draft-ietf-sip-saml-02.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt>=20
0930
eTags For Notification
Aki Niemi
draft-ietf-sip-subnot-etags-00.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00.txt>

0950
Domain Certs
Vijay Gurbani
Scott Lawrence
draft-gurbani-sip-domain-certs-05.txt
<http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs-05.t
xt>=20
1020
Privacy Guidelines
Mayumi Munakata
draft-munakata-sip-privacy-guideline-00.txt
<http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-guidelin
e-00.txt>=20
1040
Privacy Enhancements
Mayumi Munakata
draft-munakata-sip-privacy-new-00.txt
<http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new-00.t
xt>=20
1100
Certificate Authentication
Sumanth Channabasappa
draft-dotson-sip-certificate-auth-03.txt
<http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-auth-0
3.txt>=20
=09
=09
=09
=09



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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=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3086" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; khtml-nbsp-mode: space; =
khtml-line-break: after-white-space">
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I think we should put a couple of "do we =
charter" items on=20
the agenda:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>These are the mime body handling draft from =
Gonzalo, and=20
the proposed INFO handling draft from Eric.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>So far these are the only two that I can =
identify as having=20
had discussion and for which we do not have slots.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Not sure whether we need a slot for corrections =
or not - I=20
think Robert was suggesting an ad hoc somewhere to progress=20
this.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>regards</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Keith</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=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> Dean Willis=20
  [mailto:dean.willis@softarmor.com] <BR><B>Sent:</B> Sunday, July 01, =
2007 2:51=20
  AM<BR><B>To:</B> IETF SIP<BR><B>Cc:</B> Cullen Jennings; DRAGE, Keith=20
  (Keith)<BR><B>Subject:</B> Draft Agenda for SIP at IETF 69 in=20
  Chicago<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV>Here's the current draft agenda for SIP at IETF 69. I'm sure it =
is=20
  incomplete. Changes will be posted to:</DIV>
  <DIV><A=20
  =
href=3D"http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69">=
http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69</A></DIV>=

  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <P style=3D"MARGIN: 0px 0px 2px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
  size=3D6><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
23px">Draft SIP Agenda=20
  IETF 69</SPAN></FONT></P>
  <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
  size=3D4><SPAN class=3DApple-style-span style=3D"FONT-SIZE: 16px"><BR=20
  class=3Dkhtml-block-placeholder></SPAN></FONT></P>
  <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
  size=3D5><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><B>Session 1,=20
  Monday July 23 0900-1130 Grand Ballroom</B></SPAN></FONT></P>
  <TABLE=20
  style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 0px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 0px; PADDING-BOTTOM: 2px; =
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1px solid; BACKGROUND-COLOR: #ffffff"=20
  cellSpacing=3D0 cellPadding=3D0>
    <TBODY>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Start=20
        Time</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
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0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px"><B>Topic</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Discussion =

        Lead</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Reading=20
        List</B></SPAN></FONT></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0900</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Agenda Bash=20
        and Status</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">Chairs</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">This=20
        document</SPAN></FONT></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0915</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">SIPS=20
        WGLC</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Francois=20
        Audet</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt"><=
FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-ietf-sip-sips-05.txt</SPAN></FONT></A></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0945</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">Outbound</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Rohan=20
        Mahy</SPAN></FONT></DIV>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Cullen=20
        Jennings</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.tx=
t"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-ietf-sip-outbound-09.txt</SPAN></FONT></A></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1015</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Location=20
        Conveyance WGLC</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">James=20
        Polk</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conve=
yance-07.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-ietf-sip-location-conveyance-07.txt</SPAN></FONT></A></DIV></=
TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1045</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Delivering=20
        R-URI and Parameters to UA</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Jonathan=20
        Rosenberg</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-=
route-01.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-rosenberg-sip-ua-loose-route-01.txt</SPAN></FONT></A></DIV></=
TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1130</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">End of=20
        Session</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
        class=3DApple-style-span style=3D"LINE-HEIGHT: =
19px"><BR></SPAN></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
        class=3DApple-style-span=20
    style=3D"LINE-HEIGHT: =
19px"><BR></SPAN></DIV></TD></TR></TBODY></TABLE>
  <P style=3D"MARGIN: 0px 0px 5px; LINE-HEIGHT: 19px"><FONT =
class=3DApple-style-span=20
  face=3DHelvetica size=3D3><SPAN class=3DApple-style-span =
style=3D"FONT-SIZE: 12px"><BR=20
  class=3Dkhtml-block-placeholder></SPAN></FONT></P>
  <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
  size=3D5><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><B>Session 2,=20
  Tuesday July 24 0900-1130 Grand Ballroom</B></SPAN></FONT></P>
  <TABLE=20
  style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 0px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 0px; PADDING-BOTTOM: 2px; =
BORDER-LEFT: #808080 1px solid; PADDING-TOP: 2px; BORDER-BOTTOM: #808080 =
1px solid; BACKGROUND-COLOR: #ffffff"=20
  cellSpacing=3D0 cellPadding=3D0>
    <TBODY>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Start=20
        Time</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px"><B>Topic</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Discussion =

        Lead</B></SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Reading=20
        List</B></SPAN></FONT></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0900</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Agenda Bash=20
        and Status</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">Chairs</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">This=20
        document</SPAN></FONT></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0905</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">SAML</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Hannes=20
        Tschofenig</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt"><=
FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-ietf-sip-saml-02.txt</SPAN></FONT></A></DIV></TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0930</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">eTags For=20
        Notification</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Aki=20
        Niemi</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-0=
0.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-ietf-sip-subnot-etags-00.txt</SPAN></FONT></A></DIV></TD></TR=
>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">0950</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Domain=20
        Certs</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Vijay=20
        Gurbani</SPAN></FONT></DIV>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Scott=20
        Lawrence</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-cert=
s-05.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-gurbani-sip-domain-certs-05.txt</SPAN></FONT></A></DIV></TD><=
/TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1020</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Privacy=20
        Guidelines</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Mayumi=20
        Munakata</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-gu=
ideline-00.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-munakata-sip-privacy-guideline-00.txt</SPAN></FONT></A></DIV>=
</TD></TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1040</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Privacy=20
        Enhancements</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Mayumi=20
        Munakata</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-ne=
w-00.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-munakata-sip-privacy-new-00.txt</SPAN></FONT></A></DIV></TD><=
/TR>
    <TR>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span=20
        style=3D"FONT-SIZE: 12px">1100</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Certificate=20
        Authentication</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
        size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Sumanth=20
        Channabasappa</SPAN></FONT></DIV></TD>
      <TD=20
      style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
      vAlign=3Dcenter>
        <DIV style=3D"MARGIN: 0px"><A=20
        =
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-=
auth-03.txt"><FONT=20
        class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
        class=3DApple-style-span=20
        style=3D"FONT-SIZE: =
12px">draft-dotson-sip-certificate-auth-03.txt</SPAN></FONT></A></DIV></T=
D></TR></TBODY></TABLE>
  <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
  class=3DApple-style-span style=3D"LINE-HEIGHT: 19px"><BR></SPAN></DIV>
  <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
  size=3D4><SPAN class=3DApple-style-span style=3D"FONT-SIZE: 16px"><BR=20
  class=3Dkhtml-block-placeholder></SPAN></FONT></DIV>
  <DIV style=3D"MARGIN: 0px; LINE-HEIGHT: 13px"><BR=20
  class=3Dkhtml-block-placeholder></DIV></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C7BC7B.36EF07B8--



--===============1497727174==
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1497727174==--





From sip-bounces@ietf.org Mon Jul 02 09:56:25 2007
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To: IETF-Announce <ietf-announce@ietf.org>
From: The IESG <iesg-secretary@ietf.org>
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Cc: sip@ietf.org
Subject: [Sip] Last Call: draft-ietf-sip-gruu (Obtaining and Using Globally 
 Routable User Agent (UA) URIs (GRUU) in the Session Initiation 
 Protocol (SIP)) to Proposed Standard 
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The IESG has received a request from the Session Initiation Protocol WG 
(sip) to consider the following document:

- 'Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in 
   the Session Initiation Protocol (SIP) '
   <draft-ietf-sip-gruu-14.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2007-07-16. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-14.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=11271&rfc_flag=0



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



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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] RE: Draft Agenda for SIP at IETF 69 in Chicago
Date: Mon, 2 Jul 2007 13:31:19 -0500
To: Christer Holmberg (JO/LMF) <christer.holmberg@ericsson.com>
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On Jul 2, 2007, at 4:21 AM, Christer Holmberg (JO/LMF) wrote:

> Hi,
>
> Regarding the loose route issue, I still haven't received any reply  
> on whether the problem can be solved with the P-Called-Party header  
> - in which case there would be no problem to discuss.
>

well, whether it can or can't sounds like something to discuss . . .

--
Dean



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Subject: Re: [Sip] RE: Draft Agenda for SIP at IETF 69 in Chicago
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At 02:32 AM 7/2/2007, DRAGE, Keith \(Keith\) wrote:
>Content-class: urn:content-classes:message
>Content-Type: multipart/alternative;
>         boundary="----_=_NextPart_001_01C7BC7B.36EF07B8"
>
>I think we should put a couple of "do we charter" items on the agenda:

The new draft agenda is here
http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69

I'll be nice and give up 15 of my allocated 30 minutes to these other 
efforts, as I think I'll only need 15min to discuss my 2 IDs

>
>These are the mime body handling draft from Gonzalo, and the 
>proposed INFO handling draft from Eric.
>
>So far these are the only two that I can identify as having had 
>discussion and for which we do not have slots.
>
>Not sure whether we need a slot for corrections or not - I think 
>Robert was suggesting an ad hoc somewhere to progress this.
>
>regards
>
>Keith


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Subject: [Sip] SIP Identity using Media Path
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As many of you are aware, the signature created by SIP-Identity
(RFC4474) breaks if an SBC or B2BUA modifies the SDP in the SIP body.
Typically these modifications include changing the m/c lines (to
direct the RTP media through the SBC itself, or through a transcoder),
or performing other adjustments of the SDP to interwork with bugs and
features of other SIP networks or other endpoints.


I just submitted a draft, "SIP Identity using Media Path", which
describes a mechanism that allows RFC4474-like signatures and also
allows SBCs and B2BUAs to modify the message's SDP.  Links to HTML
and plain text versions of the Internet Draft are below.

Abstract:

   The existing SIP identity mechanism (RFC4474) creates a signature
   over the SIP body, including the entire SDP.  As part of their normal
   operation, Session Border Controllers (SBCs) and SIP Back-to-Back
   User Agents (B2BUAs) modify various fields in the SDP, breaking the
   signature.

   This document defines a new mechanism to securely identify the
   originator of a SIP message while also allowing modification of the
   SDP by SBCs and B2BUAs.  This new mechanism creates a signature over
   certain SIP headers and certain SDP lines.  Proof of identity over
   the media path using DTLS, TLS, HIP, and an extension to ICE are
   described.

Please send comments on this draft to the SIP mailing list at
sip@ietf.org.

-d

-----

HTML version: 
http://tinyurl.com/25539z
http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-sip-identity-med
ia-00.html

plain text version: 
http://tinyurl.com/2gldnv
http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-sip-identity-med
ia-00.txt



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here's the latest:

SIP Agenda IETF 69

Session 1, Monday July 23 0900-1130 Grand Ballroom
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0915
SIPS WGLC
Francois Audet
draft-ietf-sip-sips-05.txt
0935
Outbound
Rohan Mahy
Cullen Jennings
draft-ietf-sip-outbound-09.txt
1005
Delivering R-URI and Parameters to UA
Jonathan Rosenberg
draft-rosenberg-sip-ua-loose-route-01.txt
1040
Resource Priority Header Issues
James Polk
draft-polk-sip-rph-in-responses-00
draft-polk-sip-rph-new-namespaces-01.txt

1100
INFO And What To Do About It
Eric Burger

1115
MIME Body Handling
Gonzalo Camairllo
draft-camarillo-sip-body-handling-01.txt
1130
End of Session



Session 2, Tuesday July 24 0900-1130 Grand Ballroom
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0905
Fork Loop Fix and Corrections
Robert Sparks
draft-ietf-sip-fork-loop-fix-05
draft-drage-sip-essential-correction-01

0925
SAML
Hannes Tschofenig
draft-ietf-sip-saml-02.txt
0945
eTags For Notification
Aki Niemi
draft-ietf-sip-subnot-etags-00.txt
1000
Domain Certs
Vijay Gurbani
Scott Lawrence
draft-gurbani-sip-domain-certs-05.txt
1020
Privacy Guidelines
Mayumi Munakata
draft-munakata-sip-privacy-guideline-00.txt
1040
Privacy Enhancements
Mayumi Munakata
draft-munakata-sip-privacy-new-00.txt
1100
Certificate Authentication
Steve Dotson
draft-dotson-sip-certificate-auth-03.txt


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
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class=3D"khtml-block-placeholder"></DIV><DIV>here's the =
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margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"6"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
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0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
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size=3D"5"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
16.8px;"><B>Session 1, Monday July 23 0900-1130 Grand =
Ballroom</B></SPAN></FONT></P><TABLE cellspacing=3D"0" cellpadding=3D"0" =
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#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Start =
Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0915</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">SIPS WGLC</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Francois Audet</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-sips-05.txt</SPAN></FONT></A></DIV></TD></TR><TR><=
TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0935</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Outbound</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Rohan Mahy</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Cullen Jennings</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt=
"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-outbound-09.txt</SPAN></FONT></A></DIV></TD></TR><=
TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1005</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Delivering R-URI and Parameters to =
UA</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Jonathan Rosenberg</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-r=
oute-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-rosenberg-sip-ua-loose-route-01.txt</SPAN></FONT></A></DIV>=
</TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: =
0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px =
1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; padding: =
0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1040</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Resource Priority Header Issues</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">James Polk</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-response=
s-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-in-responses-00</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespa=
ces-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-new-namespaces-01.txt</SPAN></FONT></A></P></T=
D></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1100</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">INFO And What To Do About =
It</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Eric Burger</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12.7px/normal Helvetica; =
min-height: 16px; "><SPAN class=3D"Apple-style-span" style=3D"line-height:=
 19px;"><BR></SPAN></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1115</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">MIME Body Handling</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Gonzalo Camairllo</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handl=
ing-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-camarillo-sip-body-handling-01.txt</SPAN></FONT></A></DIV><=
/TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1130</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">End of Session</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12.7px/normal Helvetica; min-height: 16px; "><SPAN =
class=3D"Apple-style-span" style=3D"line-height: =
19px;"><BR></SPAN></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12.7px/normal Helvetica; =
min-height: 16px; "><SPAN class=3D"Apple-style-span" style=3D"line-height:=
 19px;"><BR></SPAN></DIV></TD></TR></TBODY></TABLE><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><BR class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><P =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 5px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"5"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
16.8px;"><B>Session 2, Tuesday July 24 0900-1130 Grand =
Ballroom</B></SPAN></FONT></P><TABLE cellspacing=3D"0" cellpadding=3D"0" =
style=3D"background-color: #ffffff; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 2.0px 0.0px 2.0px 0.0px"><TBODY><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Start =
Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0905</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Fork Loop Fix and Corrections</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Robert Sparks</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-fork-loop-fix-0=
5.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-fork-loop-fix-05</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-drage-sip-essential-corr=
ection-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-drage-sip-essential-correction-01</SPAN></FONT></A></P></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0925</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">SAML</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Hannes =
Tschofenig</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-saml-02.txt</SPAN></FONT></A></DIV></TD></TR><TR><=
TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0945</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">eTags For =
Notification</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width:=
 112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Aki Niemi</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00=
.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN=
 class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-subnot-etags-00.txt</SPAN></FONT></A></DIV></TD></=
TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1000</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Domain Certs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Vijay Gurbani</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Scott Lawrence</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs=
-05.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-gurbani-sip-domain-certs-05.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1020</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Privacy =
Guidelines</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Mayumi Munakata</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-gui=
deline-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-munakata-sip-privacy-guideline-00.txt</SPAN></FONT></A></DI=
V></TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: =
0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px =
1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; padding: =
0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1040</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Privacy Enhancements</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Mayumi Munakata</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new=
-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-munakata-sip-privacy-new-00.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1100</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Certificate =
Authentication</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Steve Dotson</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 259.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-a=
uth-03.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-dotson-sip-certificate-auth-03.txt</SPAN></FONT></A></DIV><=
/TD></TR></TBODY></TABLE><DIV><BR =
class=3D"khtml-block-placeholder"></DIV></BODY></HTML>=

--Apple-Mail-6--1061233573--



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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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	"Dean Willis" <dean.willis@softarmor.com>, "IETF SIP" <sip@ietf.org>
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Hi,
=20
Regarding the loose route issue, I still haven't received any reply on =
whether the problem can be solved with the P-Called-Party header - in =
which case there would be no problem to discuss.
=20
Regards,
=20
Christer
=20


________________________________

	From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
	Sent: 2. hein=E4kuuta 2007 10:33
	To: Dean Willis; IETF SIP
	Cc: Cullen Jennings
	Subject: [Sip] RE: Draft Agenda for SIP at IETF 69 in Chicago
=09
=09
	I think we should put a couple of "do we charter" items on the agenda:
	=20
	These are the mime body handling draft from Gonzalo, and the proposed =
INFO handling draft from Eric.
	=20
	So far these are the only two that I can identify as having had =
discussion and for which we do not have slots.
	=20
	Not sure whether we need a slot for corrections or not - I think Robert =
was suggesting an ad hoc somewhere to progress this.
	=20
	regards
	=20
	Keith


________________________________

		From: Dean Willis [mailto:dean.willis@softarmor.com]=20
		Sent: Sunday, July 01, 2007 2:51 AM
		To: IETF SIP
		Cc: Cullen Jennings; DRAGE, Keith (Keith)
		Subject: Draft Agenda for SIP at IETF 69 in Chicago
	=09
	=09
		Here's the current draft agenda for SIP at IETF 69. I'm sure it is =
incomplete. Changes will be posted to:
		http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69


		Draft SIP Agenda IETF 69

	=09
	=09

		Session 1, Monday July 23 0900-1130 Grand Ballroom

Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0915
SIPS WGLC
Francois Audet
draft-ietf-sip-sips-05.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt>=20
0945
Outbound
Rohan Mahy
Cullen Jennings
draft-ietf-sip-outbound-09.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt>=20
1015
Location Conveyance WGLC
James Polk
draft-ietf-sip-location-conveyance-07.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-0=
7.txt>=20
1045
Delivering R-URI and Parameters to UA
Jonathan Rosenberg
draft-rosenberg-sip-ua-loose-route-01.txt =
<http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-route-0=
1.txt>=20
1130
End of Session





	=09
	=09

		Session 2, Tuesday July 24 0900-1130 Grand Ballroom

Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0905
SAML
Hannes Tschofenig
draft-ietf-sip-saml-02.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt>=20
0930
eTags For Notification
Aki Niemi
draft-ietf-sip-subnot-etags-00.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00.txt> =

0950
Domain Certs
Vijay Gurbani
Scott Lawrence
draft-gurbani-sip-domain-certs-05.txt =
<http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs-05.tx=
t>=20
1020
Privacy Guidelines
Mayumi Munakata
draft-munakata-sip-privacy-guideline-00.txt =
<http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-guideline=
-00.txt>=20
1040
Privacy Enhancements
Mayumi Munakata
draft-munakata-sip-privacy-new-00.txt =
<http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new-00.tx=
t>=20
1100
Certificate Authentication
Sumanth Channabasappa
draft-dotson-sip-certificate-auth-03.txt =
<http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-auth-03=
.txt>=20
	=09
	=09
	=09
	=09



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	charset="iso-8859-1"
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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 6.00.2800.1595" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; khtml-nbsp-mode: space; =
khtml-line-break: after-white-space">
<DIV><SPAN class=3D345132009-02072007><FONT face=3DArial color=3D#0000ff =

size=3D2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=3D345132009-02072007><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D345132009-02072007><FONT face=3DArial color=3D#0000ff =

size=3D2>Regarding the loose route issue, I still haven't received any =
reply on=20
whether the problem can be solved with the P-Called-Party header - in =
which case=20
there&nbsp;would be&nbsp;no problem to discuss.</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D345132009-02072007>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D345132009-02072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D345132009-02072007>Christer</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><BR></DIV>
<BLOCKQUOTE dir=3Dltr=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> DRAGE, Keith (Keith)=20
  [mailto:drage@alcatel-lucent.com] <BR><B>Sent:</B> 2. hein=E4kuuta =
2007=20
  10:33<BR><B>To:</B> Dean Willis; IETF SIP<BR><B>Cc:</B> Cullen=20
  Jennings<BR><B>Subject:</B> [Sip] RE: Draft Agenda for SIP at IETF 69 =
in=20
  Chicago<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>I think we should put a couple of "do we =
charter" items=20
  on the agenda:</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>These are the mime body handling draft from =
Gonzalo, and=20
  the proposed INFO handling draft from Eric.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>So far these are the only two that I can =
identify as=20
  having had discussion and for which we do not have =
slots.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>Not sure whether we need a slot for =
corrections or not -=20
  I think Robert was suggesting an ad hoc somewhere to progress=20
  this.</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>regards</FONT></SPAN></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D564460300-02072007><FONT =
face=3DArial=20
  color=3D#0000ff size=3D2>Keith</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE dir=3Dltr=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> Dean Willis=20
    [mailto:dean.willis@softarmor.com] <BR><B>Sent:</B> Sunday, July 01, =
2007=20
    2:51 AM<BR><B>To:</B> IETF SIP<BR><B>Cc:</B> Cullen Jennings; DRAGE, =
Keith=20
    (Keith)<BR><B>Subject:</B> Draft Agenda for SIP at IETF 69 in=20
    Chicago<BR></FONT><BR></DIV>
    <DIV></DIV>
    <DIV>Here's the current draft agenda for SIP at IETF 69. I'm sure it =
is=20
    incomplete. Changes will be posted to:</DIV>
    <DIV><A=20
    =
href=3D"http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69">=
http://www.softarmor.com/mediawiki/index.php/SIP_Agenda_IETF_69</A></DIV>=

    <DIV><BR class=3Dkhtml-block-placeholder></DIV>
    <P style=3D"MARGIN: 0px 0px 2px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
    size=3D6><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
23px">Draft SIP Agenda=20
    IETF 69</SPAN></FONT></P>
    <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
    size=3D4><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><BR=20
    class=3Dkhtml-block-placeholder></SPAN></FONT></P>
    <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
    size=3D5><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><B>Session 1,=20
    Monday July 23 0900-1130 Grand Ballroom</B></SPAN></FONT></P>
    <TABLE=20
    style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 0px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 0px; PADDING-BOTTOM: 2px; =
BORDER-LEFT: #808080 1px solid; PADDING-TOP: 2px; BORDER-BOTTOM: #808080 =
1px solid; BACKGROUND-COLOR: #ffffff"=20
    cellSpacing=3D0 cellPadding=3D0>
      <TBODY>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Start=20
          Time</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px"><B>Topic</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: =
12px"><B>Discussion=20
          Lead</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Reading=20
          List</B></SPAN></FONT></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0900</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Agenda=20
          Bash and Status</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">Chairs</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">This=20
          document</SPAN></FONT></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0915</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">SIPS=20
          WGLC</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Francois=20
          Audet</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt"><=
FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-ietf-sip-sips-05.txt</SPAN></FONT></A></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0945</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">Outbound</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Rohan=20
          Mahy</SPAN></FONT></DIV>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Cullen=20
          Jennings</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.tx=
t"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-ietf-sip-outbound-09.txt</SPAN></FONT></A></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1015</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Location=20
          Conveyance WGLC</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">James=20
          Polk</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conve=
yance-07.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-ietf-sip-location-conveyance-07.txt</SPAN></FONT></A></DIV></=
TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1045</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Delivering=20
          R-URI and Parameters to UA</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Jonathan=20
          Rosenberg</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-=
route-01.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-rosenberg-sip-ua-loose-route-01.txt</SPAN></FONT></A></DIV></=
TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1130</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 229px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">End of=20
          Session</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 119px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
          class=3DApple-style-span style=3D"LINE-HEIGHT: =
19px"><BR></SPAN></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 244px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
          class=3DApple-style-span=20
      style=3D"LINE-HEIGHT: =
19px"><BR></SPAN></DIV></TD></TR></TBODY></TABLE>
    <P style=3D"MARGIN: 0px 0px 5px; LINE-HEIGHT: 19px"><FONT=20
    class=3DApple-style-span face=3DHelvetica size=3D3><SPAN =
class=3DApple-style-span=20
    style=3D"FONT-SIZE: 12px"><BR =
class=3Dkhtml-block-placeholder></SPAN></FONT></P>
    <P style=3D"MARGIN: 0px 0px 5px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
    size=3D5><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><B>Session 2,=20
    Tuesday July 24 0900-1130 Grand Ballroom</B></SPAN></FONT></P>
    <TABLE=20
    style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 0px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 0px; PADDING-BOTTOM: 2px; =
BORDER-LEFT: #808080 1px solid; PADDING-TOP: 2px; BORDER-BOTTOM: #808080 =
1px solid; BACKGROUND-COLOR: #ffffff"=20
    cellSpacing=3D0 cellPadding=3D0>
      <TBODY>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Start=20
          Time</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px"><B>Topic</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: =
12px"><B>Discussion=20
          Lead</B></SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px; TEXT-ALIGN: center"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span style=3D"FONT-SIZE: 12px"><B>Reading=20
          List</B></SPAN></FONT></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0900</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Agenda=20
          Bash and Status</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">Chairs</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">This=20
          document</SPAN></FONT></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0905</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">SAML</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Hannes=20
          Tschofenig</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt"><=
FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-ietf-sip-saml-02.txt</SPAN></FONT></A></DIV></TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0930</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">eTags For=20
          Notification</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Aki=20
          Niemi</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-0=
0.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-ietf-sip-subnot-etags-00.txt</SPAN></FONT></A></DIV></TD></TR=
>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">0950</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Domain=20
          Certs</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Vijay=20
          Gurbani</SPAN></FONT></DIV>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Scott=20
          Lawrence</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-cert=
s-05.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-gurbani-sip-domain-certs-05.txt</SPAN></FONT></A></DIV></TD><=
/TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1020</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Privacy=20
          Guidelines</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Mayumi=20
          Munakata</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-gu=
ideline-00.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-munakata-sip-privacy-guideline-00.txt</SPAN></FONT></A></DIV>=
</TD></TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1040</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Privacy=20
          Enhancements</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Mayumi=20
          Munakata</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-ne=
w-00.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-munakata-sip-privacy-new-00.txt</SPAN></FONT></A></DIV></TD><=
/TR>
      <TR>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 64px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">1100</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 146px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span=20
          style=3D"FONT-SIZE: 12px">Certificate=20
        Authentication</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 149px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
          size=3D3><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
12px">Sumanth=20
          Channabasappa</SPAN></FONT></DIV></TD>
        <TD=20
        style=3D"BORDER-RIGHT: #808080 1px solid; PADDING-RIGHT: 2px; =
BORDER-TOP: #808080 1px solid; PADDING-LEFT: 2px; PADDING-BOTTOM: 0px; =
MARGIN: 0px; BORDER-LEFT: #808080 1px solid; WIDTH: 259px; PADDING-TOP: =
0px; BORDER-BOTTOM: #808080 1px solid"=20
        vAlign=3Dcenter>
          <DIV style=3D"MARGIN: 0px"><A=20
          =
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-=
auth-03.txt"><FONT=20
          class=3DApple-style-span face=3DHelvetica size=3D3><SPAN=20
          class=3DApple-style-span=20
          style=3D"FONT-SIZE: =
12px">draft-dotson-sip-certificate-auth-03.txt</SPAN></FONT></A></DIV></T=
D></TR></TBODY></TABLE>
    <DIV style=3D"MIN-HEIGHT: 16px; MARGIN: 0px; FONT: 12px =
Helvetica"><SPAN=20
    class=3DApple-style-span style=3D"LINE-HEIGHT: =
19px"><BR></SPAN></DIV>
    <DIV style=3D"MARGIN: 0px"><FONT class=3DApple-style-span =
face=3DHelvetica=20
    size=3D4><SPAN class=3DApple-style-span style=3D"FONT-SIZE: =
16px"><BR=20
    class=3Dkhtml-block-placeholder></SPAN></FONT></DIV>
    <DIV style=3D"MARGIN: 0px; LINE-HEIGHT: 13px"><BR=20
    =
class=3Dkhtml-block-placeholder></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></=
HTML>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1428730687==--





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Subject: RE: [Sip] SIP Identity using Media Path
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Hi Dan,

Chapter 9 says:

"in order for the mechanism to work, SBC-type-of-entities must permit =
DTLS, TLS, ICE, or HIP messages to be exchanged in the media path."

A small question for clarification: at what point must this exchange =
(two-way, I assume) work? As soon as the UAS has received the INVITE?

Regards,

Christer


> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]=20
> Sent: 2. hein=E4kuuta 2007 21:56
> To: 'SIP Working Group'
> Cc: ietf-rtpsec@imc.org
> Subject: [Sip] SIP Identity using Media Path
>=20
> As many of you are aware, the signature created by SIP-Identity
> (RFC4474) breaks if an SBC or B2BUA modifies the SDP in the SIP body.
> Typically these modifications include changing the m/c lines=20
> (to direct the RTP media through the SBC itself, or through a=20
> transcoder), or performing other adjustments of the SDP to=20
> interwork with bugs and features of other SIP networks or=20
> other endpoints.
>=20
>=20
> I just submitted a draft, "SIP Identity using Media Path",=20
> which describes a mechanism that allows RFC4474-like=20
> signatures and also allows SBCs and B2BUAs to modify the=20
> message's SDP.  Links to HTML and plain text versions of the=20
> Internet Draft are below.
>=20
> Abstract:
>=20
>    The existing SIP identity mechanism (RFC4474) creates a signature
>    over the SIP body, including the entire SDP.  As part of=20
> their normal
>    operation, Session Border Controllers (SBCs) and SIP Back-to-Back
>    User Agents (B2BUAs) modify various fields in the SDP, breaking the
>    signature.
>=20
>    This document defines a new mechanism to securely identify the
>    originator of a SIP message while also allowing modification of the
>    SDP by SBCs and B2BUAs.  This new mechanism creates a=20
> signature over
>    certain SIP headers and certain SDP lines.  Proof of identity over
>    the media path using DTLS, TLS, HIP, and an extension to ICE are
>    described.
>=20
> Please send comments on this draft to the SIP mailing list at=20
> sip@ietf.org.
>=20
> -d
>=20
> -----
>=20
> HTML version:=20
> http://tinyurl.com/25539z
> http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-si
> p-identity-med
> ia-00.html
>=20
> plain text version:=20
> http://tinyurl.com/2gldnv
> http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-si
> p-identity-med
> ia-00.txt
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Mon Jul 02 19:05:52 2007
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From: "Dan Wing" <dwing@cisco.com>
To: "'Christer Holmberg \(JO/LMF\)'" <christer.holmberg@ericsson.com>,
	<sip@ietf.org>
Subject: RE: [Sip] SIP Identity using Media Path
Date: Mon, 2 Jul 2007 16:04:54 -0700
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> Chapter 9 says:
> 
> "in order for the mechanism to work, SBC-type-of-entities 
> must permit DTLS, TLS, ICE, or HIP messages to be exchanged 
> in the media path."
> 
> A small question for clarification: at what point must this 
> exchange (two-way, I assume) work? As soon as the UAS has 
> received the INVITE?

Yes.

The DTLS, TLS, ICE, or HIP exchange would need to occur
prior to displaying Caller-ID-like information to the
user -- otherwise, an attacker could spoof that Caller-ID
information.  It seems defective, to me, to display
Caller-ID information that was spoofed to only later
learn (after doing DTLS, TLS, ICE, or HIP) that the 
calling party had lied about their identity.  

There is some protection against such an attack that could
reduce this risk (the Date: header is signed, so the attacker
would need to obtain a relatively recent SIP request) -- that
protection might be sufficient in networks that block exchange 
of packets prior to 200 Ok.  If an attacker was able to
get ahold of such a SIP request, you would know it when the
attacker failed the DTLS, TLS, ICE, or HIP exchange, which
I suppose is better than nothing.

-d


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From sip-bounces@ietf.org Mon Jul 02 22:38:10 2007
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Subject: [Sip] Security issues with ietf-sip-loop-fix
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(this email with my AD hat on)

The type of SIP loops described in ietf-sip-loop-fix can be roughly  
classified into two categories. Loops that had happened because an  
accidental registration that causes a loop, and loops that happened  
due to malicious intent.

This draft attempted to deals with both. The abstract describes it as

   This document normatively updates RFC 3261, the Session Initiation
    Protocol (SIP), to address a security vulnerability identified in  
SIP
    proxy behavior.  This vulnerability enables an attack against SIP
    networks where a small number of legitimate, even authorized, SIP
    requests can stimulate massive amounts of proxy-to-proxy traffic.

During IETF LC, substantial comment came up about the security of  
this draft not really working in the case of malicious loops. A  
thread ensues that convinced many people on the thread that this  
draft did not really help with the malicious loops. (Not 100% sure  
here but I suspect that the authors and chairs were all convinced of  
this). I will let others explain the issues that came up as we decide  
what to do next.

I think that some of issue was discussed while the working group was  
considering this draft - however, I suspect that many participants  
did not understand how easily this security would be defeated. I for  
one significantly underestimated the how easy it was for a malicious  
attacker to cause this problem. I apologize to the folks that  
attempted to explain it to me and I just failed to "get it".

Anyways, we need to figure out what to do next with this draft. There  
are basically two paths forward that I see:

1) reduce the scope so that it only deals with accidental loops and  
provides no protection against malicious loops. I don't think this  
would change any of the mechanisms in the draft but it would probably  
change lots of the text around what security claims the draft made.

2) try to extend or replace the mechanism with one that does protect  
against malicious attack. A strong candidate to consider would be the  
work in draft-sparks-sipping-max-breadth.

If we go forward with path 1, the value of this draft is probably not  
super high because Max-Forwards deals with the the accidental loops  
in most cases  (but not all cases). If we go with path 2, the WG will  
need make sure it OK with the solution. I'm certainly open to other  
paths forward if someone has a good idea.

What I would like to do now is start some discussion about what the  
working group would like to do with this draft.

Thanks, Cullen






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From sip-bounces@ietf.org Tue Jul 03 05:21:34 2007
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Date: Tue, 03 Jul 2007 17:21:01 +0800
From: Linyi Tian <tianlinyi@huawei.com>
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Subject: [Sip] Comments and questions to
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Hi, all
(1) On section "5.  Conference Server Procedures"
Many conference servers have restriction to maximum number of attendees for conference. If the users of 'recipient-list' exceeds the limitation, what status code should be returned?

---------------------------------------
7.  Security Considerations
   This document discusses setup of SIP conferences using a request-contained URI-list.  Both conferencing and URI-lists services have specific security requirements which will be summarized here. Conferences generally have authorization rules about who can or cannot join a conference, what type of media can or cannot be used, etc.  This information is used by the focus to admit or deny participation in a conference.  It is RECOMMENDED that these types of authorization rules be used to provide security for a SIP conference.
----------------------------------------
(2) When and how the authorization rules are specified and applied to the conference? Before the conference creation seems not possible since Conf-URI is not assigned. If specified after the conference creation, it seems there is a risk after conf is created but rules are not ready. Some attendees may do something bad which is out of control during that period. 

Best Regards,
Linyi





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Hello,

just my though about the draft, related to the requirement=20
	"The client transaction SHOULD start timer D when it enters the
"Completed" state for any reason, with a value of at least 32 seconds
for unreliable transports, and a value of zero seconds for reliable
transports":=20

For client transactions, wouldn't it be better that the timer D is set
to 32s even for reliable transport after receiving 200-299 response, so
the transaction can collect also responses to a forked INVITE? This
would align the INVITE transaction with the UAC Core behaviour, which
according to "RFC3261 13.2.2.4 2xx Responses" considers the INVITE
transaction completed 64*T1 seconds after the reception of the first 2xx
response.
Also the 200-299 is retransmitted end-to-end, so the reliability of the
transport to the first hop does not really mean that the response or ACK
is sent reliably all the way from UAS to UAC, respectivelly from UAC to
UAS.

=20
David Bohaty
System Architect
=20
SITRONICS Telecom Solutions, Czech Republic a.s.=20
Tel.: +420 226 532 578, Fax: +420 241 480 899
BB Centrum - Beta, Vyskocilova 1481/4, 140 00  Praha 4, Czech Republic
www.sitronicsts.com


-----Original Message-----
From: Robert Sparks [mailto:rjsparks@estacado.net]=20
Sent: 28 June 2007 04:35
To: SIP IETF
Subject: [Sip] Changes to the INVITE transaction to address bug 769

Everyone -

I've put together a proposal for addressing the spec bug captured at
http://bugs.sipit.net/show_bug.cgi?id=3D769

Please read through
http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
and comment.

RjS


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From sip-bounces@ietf.org Tue Jul 03 06:19:07 2007
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See flows sections 6.3 and 6.4.

F13' shows an INVITE to Bob's PC but the INVITE message shows:
INVITE sip:bob@bobphone.example.com

it should be:
INVITE sip:bob@bobpc.example.com



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From sip-bounces@ietf.org Tue Jul 03 09:32:31 2007
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Subject: RE: [Sip] SIP Identity using Media Path
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Hi,=20

>>Chapter 9 says:
>>=20
>>"in order for the mechanism to work, SBC-type-of-entities must permit=20
>>DTLS, TLS, ICE, or HIP messages to be exchanged in the media path."
>>=20
>>A small question for clarification: at what point must this exchange=20
>>(two-way, I assume) work? As soon as the UAS has received the INVITE?
>=20
>Yes.

As we know from our RTPSEC discussions, one of the issues of using
SBC-type-of-entities (or gates) is that you MAY NOT have end-to-end
media plane connectivity as soon as the INVITE has been sent.

Regards,

Christer


=20
>The DTLS, TLS, ICE, or HIP exchange would need to occur prior=20
>to displaying Caller-ID-like information to the user --=20
>otherwise, an attacker could spoof that Caller-ID=20
>information.  It seems defective, to me, to display Caller-ID=20
>information that was spoofed to only later learn (after doing=20
>DTLS, TLS, ICE, or HIP) that the calling party had lied about=20
>their identity. =20
>=20
>There is some protection against such an attack that could=20
>reduce this risk (the Date: header is signed, so the attacker=20
>would need to obtain a relatively recent SIP request) -- that=20
>protection might be sufficient in networks that block=20
>exchange of packets prior to 200 Ok.  If an attacker was able=20
>to get ahold of such a SIP request, you would know it when=20
>the attacker failed the DTLS, TLS, ICE, or HIP exchange,=20
>which I suppose is better than nothing.
>=20
> -d
>=20


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From sip-bounces@ietf.org Tue Jul 03 09:50:39 2007
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Security issues with ietf-sip-loop-fix
Date: Tue, 3 Jul 2007 08:49:18 -0500
To: Cullen Jennings <fluffy@cisco.com>
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We've just submitted an update to the max-breadth draft - it greatly  
simplified the text of -00 and explores
the problem identified in the secdir review of fork-loop-fix a little.

In a nutshell, that problem is that an attacker can get a large  
number (on the order of max-forwards) of
AoRs to participate in the attack. In the real world, this is not  
hard to do. At that point, neither max-forwards
nor loop detection mitigates the damage of the attack. Furthermore,  
the damage is not inherently limited
to the services that provided the participating AoRs. The forking  
behavior can be configured such that
an innocent (perhaps not even SIP) endpoint gets a message every time  
the system forwards and forks.

The max-breadth draft as written now controls only the amplitude of  
such an attack at any given point in
time - it could be modified to control the total number of messages  
the attacked endpoint would see
(see open issue #1).

You can find the revised max-breadth draft at
http://www.nostrum.com/~rjsparks/draft-sparks-sipping-max-breadth-01.txt
until it appears in the repository. Even though it says sipping,  
please (for now) keep the discussion of
it on the SIP list.

Thanks!

RjS

On Jul 2, 2007, at 9:35 PM, Cullen Jennings wrote:

>
> (this email with my AD hat on)
>
> The type of SIP loops described in ietf-sip-loop-fix can be roughly  
> classified into two categories. Loops that had happened because an  
> accidental registration that causes a loop, and loops that happened  
> due to malicious intent.
>
> This draft attempted to deals with both. The abstract describes it as
>
>   This document normatively updates RFC 3261, the Session Initiation
>    Protocol (SIP), to address a security vulnerability identified  
> in SIP
>    proxy behavior.  This vulnerability enables an attack against SIP
>    networks where a small number of legitimate, even authorized, SIP
>    requests can stimulate massive amounts of proxy-to-proxy traffic.
>
> During IETF LC, substantial comment came up about the security of  
> this draft not really working in the case of malicious loops. A  
> thread ensues that convinced many people on the thread that this  
> draft did not really help with the malicious loops. (Not 100% sure  
> here but I suspect that the authors and chairs were all convinced  
> of this). I will let others explain the issues that came up as we  
> decide what to do next.
>
> I think that some of issue was discussed while the working group  
> was considering this draft - however, I suspect that many  
> participants did not understand how easily this security would be  
> defeated. I for one significantly underestimated the how easy it  
> was for a malicious attacker to cause this problem. I apologize to  
> the folks that attempted to explain it to me and I just failed to  
> "get it".
>
> Anyways, we need to figure out what to do next with this draft.  
> There are basically two paths forward that I see:
>
> 1) reduce the scope so that it only deals with accidental loops and  
> provides no protection against malicious loops. I don't think this  
> would change any of the mechanisms in the draft but it would  
> probably change lots of the text around what security claims the  
> draft made.
>
> 2) try to extend or replace the mechanism with one that does  
> protect against malicious attack. A strong candidate to consider  
> would be the work in draft-sparks-sipping-max-breadth.
>
> If we go forward with path 1, the value of this draft is probably  
> not super high because Max-Forwards deals with the the accidental  
> loops in most cases  (but not all cases). If we go with path 2, the  
> WG will need make sure it OK with the solution. I'm certainly open  
> to other paths forward if someone has a good idea.
>
> What I would like to do now is start some discussion about what the  
> working group would like to do with this draft.
>
> Thanks, Cullen
>
>
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org Tue Jul 03 10:04:53 2007
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From: Robert Sparks <rjsparks@estacado.net>
Subject: Re: [Sip] Changes to the INVITE transaction to address bug 769
Date: Tue, 3 Jul 2007 09:03:40 -0500
To: Bohaty David <David.Bohaty@sitronicsts.com>
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Hrmm. I think you're right, and the place where it makes a big  
difference is at
proxies in the path of the 200 that have a TCP hop on either side of  
them - we want
that client transaction to still be there for retransmissions of the  
200.

Whether the fix is as simple as setting timer D will take more  
thinking (and I want
to wait for more feedback on the rest of the draft before cracking  
this open).

I've started a list of things to work through with this item.

Thanks,

RjS


On Jul 3, 2007, at 4:30 AM, Bohaty David wrote:

> Hello,
>
> just my though about the draft, related to the requirement
> 	"The client transaction SHOULD start timer D when it enters the
> "Completed" state for any reason, with a value of at least 32 seconds
> for unreliable transports, and a value of zero seconds for reliable
> transports":
>
> For client transactions, wouldn't it be better that the timer D is set
> to 32s even for reliable transport after receiving 200-299  
> response, so
> the transaction can collect also responses to a forked INVITE? This
> would align the INVITE transaction with the UAC Core behaviour, which
> according to "RFC3261 13.2.2.4 2xx Responses" considers the INVITE
> transaction completed 64*T1 seconds after the reception of the  
> first 2xx
> response.
> Also the 200-299 is retransmitted end-to-end, so the reliability of  
> the
> transport to the first hop does not really mean that the response  
> or ACK
> is sent reliably all the way from UAS to UAC, respectivelly from  
> UAC to
> UAS.
>
>
> David Bohaty
> System Architect
>
> SITRONICS Telecom Solutions, Czech Republic a.s.
> Tel.: +420 226 532 578, Fax: +420 241 480 899
> BB Centrum - Beta, Vyskocilova 1481/4, 140 00  Praha 4, Czech Republic
> www.sitronicsts.com
>
>
> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@estacado.net]
> Sent: 28 June 2007 04:35
> To: SIP IETF
> Subject: [Sip] Changes to the INVITE transaction to address bug 769
>
> Everyone -
>
> I've put together a proposal for addressing the spec bug captured at
> http://bugs.sipit.net/show_bug.cgi?id=769
>
> Please read through
> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
> and comment.
>
> RjS
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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



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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		: Addressing Record-Route issues in the Session Initiation Protocol (SIP)
	Author(s)	: T. Froment, C. Lebel
	Filename	: draft-ietf-sip-record-route-fix-00.txt
	Pages		: 26
	Date		: 2007-7-3
	
   A typical function of a Session Initiation Protocol (SIP) Proxy is to
   set a Record-Route header on initial requests in order to make
   subsequent requests pass through it.  This header contains a SIP
   Uniform Resource Identifier (URI) indicating where and how the
   subsequent requests should be sent to reach the proxy.  Like any SIP
   URI, it can contain sip or sips schemes, IPV4 or IPV6 addresses, and
   URI parameters that could influence the routing like different
   transport parameters (UDP, TCP, SCTP...), or a compression indication
   like "comp=sigcomp".  When a proxy has to change some of those
   parameters between its incoming and outgoing interfaces (multi-homed
   proxies, transport protocol switching, sip to sips or IPV4 to IPV6
   scenarios...), the question arises on what should be put in Record-
   Route header(s).  It is just not possible to make one header having
   the characteristics of both sides at the same time.  This document
   aims to clarify these scenarios and fix bugs already identified on
   this topic; it formally recommends the use of the double Record-Route
   technique as an alternative to the current RFC3261 text, which only
   describes Record-Route rewriting solution.


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






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This is getting closer to the final agenda. If you're on the agenda  
and don't need to be, PLEASE LET ME KNOW!

--
Dean


SIP Agenda IETF 69

Session 1, Monday July 23 0900-1130 Grand Ballroom
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0915
SIPS WGLC
Francois Audet
draft-ietf-sip-sips-05.txt
0935
Outbound
Rohan Mahy
Cullen Jennings
draft-ietf-sip-outbound-09.txt
1005
Delivering R-URI and Parameters to UA
Jonathan Rosenberg
draft-rosenberg-sip-ua-loose-route-01.txt
1040
Resource Priority Header Issues
James Polk
draft-polk-sip-rph-in-responses-00
draft-polk-sip-rph-new-namespaces-01.txt

1105
MIME Body Handling
Gonzalo Camarillo
draft-camarillo-sip-body-handling-01.txt
1130
End of Session



Session 2, Tuesday July 24 0900-1130 Grand Ballroom
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0905
Fork Loop Fix and Corrections
Robert Sparks
draft-ietf-sip-fork-loop-fix-05
draft-sparks-sip-max-breadth-01
draft-drage-sip-essential-correction-01

0925
SAML
Hannes Tschofenig
draft-ietf-sip-saml-02
0945
eTags For Notification
Aki Niemi
draft-ietf-sip-subnot-etags-00.txt
1000
Domain Certs and Connection Reuse
Vijay Gurbani
Scott Lawrence
draft-gurbani-sip-domain-certs-05.txt
1025
UA-Driven Privacy
Mayumi Munakata
draft-munakata-sip-privacy-new-00.txt
1045
Certificate Authentication
Steve Dotson
draft-dotson-sip-certificate-auth-03.txt
1100
INFO Considered Harmful
Eric Burger
draft-burger-sip-info-00
1115
Media identity
Dan Wing
(waiting)
1130
End of session




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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; "><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>This is getting closer to =
the final agenda. If you're on the agenda and don't need to be, PLEASE =
LET ME KNOW!</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>--</DIV><DIV>Dean</DIV><DIV><=
BR class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 2px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"6"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 23.9px;">SIP Agenda IETF =
69</SPAN></FONT></P><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"4"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 16px;"><BR =
class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><P =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 5px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"5"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
16.8px;"><B>Session 1, Monday July 23 0900-1130 Grand =
Ballroom</B></SPAN></FONT></P><TABLE cellspacing=3D"0" cellpadding=3D"0" =
style=3D"background-color: #ffffff; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 2.0px 0.0px 2.0px 0.0px"><TBODY><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Start =
Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0915</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">SIPS WGLC</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Francois Audet</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-sips-05.txt</SPAN></FONT></A></DIV></TD></TR><TR><=
TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0935</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Outbound</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Rohan Mahy</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Cullen Jennings</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt=
"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-outbound-09.txt</SPAN></FONT></A></DIV></TD></TR><=
TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1005</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Delivering R-URI and Parameters to =
UA</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Jonathan Rosenberg</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-r=
oute-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-rosenberg-sip-ua-loose-route-01.txt</SPAN></FONT></A></DIV>=
</TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: =
0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px =
1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; padding: =
0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1040</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Resource Priority Header Issues</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">James Polk</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-response=
s-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-in-responses-00</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespa=
ces-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-new-namespaces-01.txt</SPAN></FONT></A></P></T=
D></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1105</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">MIME Body =
Handling</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Gonzalo Camarillo</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handl=
ing-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-camarillo-sip-body-handling-01.txt</SPAN></FONT></A></DIV><=
/TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1130</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">End of Session</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12.7px/normal Helvetica; min-height: 16px; "><SPAN =
class=3D"Apple-style-span" style=3D"line-height: =
19px;"><BR></SPAN></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12.7px/normal Helvetica; =
min-height: 16px; "><SPAN class=3D"Apple-style-span" style=3D"line-height:=
 19px;"><BR></SPAN></DIV></TD></TR></TBODY></TABLE><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><BR class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><P =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 5px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"5"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
16.8px;"><B>Session 2, Tuesday July 24 0900-1130 Grand =
Ballroom</B></SPAN></FONT></P><TABLE cellspacing=3D"0" cellpadding=3D"0" =
style=3D"background-color: #ffffff; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 2.0px 0.0px 2.0px 0.0px"><TBODY><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Start =
Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
214.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0905</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Fork Loop Fix and Corrections</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Robert Sparks</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-fork-loop-fix-0=
5.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-fork-loop-fix-05</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-max-breadth-0=
1.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-sparks-sip-max-breadth-01</SPAN></FONT></A><SPAN =
class=3D"Apple-style-span" style=3D"line-height: 19px;"><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px; line-height: =
normal;"> </SPAN></FONT><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px; line-height: =
normal;"><BR></SPAN></FONT></SPAN><A =
href=3D"http://www.ietf.org/internet-drafts/draft-drage-sip-essential-corr=
ection-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><FONT class=3D"Apple-style-span" =
color=3D"#4268B3">draft-drage-sip-essential-correction-01</FONT></SPAN></F=
ONT></A></P></TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; =
margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0925</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">SAML</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Hannes Tschofenig</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-saml-02</SPAN></FONT></A></DIV></TD></TR><TR><TD =
valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0945</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">eTags For =
Notification</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width:=
 112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Aki Niemi</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00=
.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN=
 class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-subnot-etags-00.txt</SPAN></FONT></A></DIV></TD></=
TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1000</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Domain Certs and Connection =
Reuse</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Vijay Gurbani</SPAN></FONT></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;">Scott =
Lawrence</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs=
-05.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-gurbani-sip-domain-certs-05.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1025</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">UA-Driven =
Privacy</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Mayumi Munakata</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new=
-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-munakata-sip-privacy-new-00.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1045</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Certificate =
Authentication</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Steve Dotson</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-a=
uth-03.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-dotson-sip-certificate-auth-03.txt</SPAN></FONT></A></DIV><=
/TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1100</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">INFO Considered =
Harmful</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
112.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Eric Burger</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-burger-sip-info-00"><FON=
T class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-burger-sip-info-00</SPAN></FONT></A></DIV></TD></TR><TR><TD=
 valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1115</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Media identity</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Dan Wing</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: =
12.7px;">(waiting)</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1130</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 214.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">End of session</SPAN></FONT></DIV></TD><TD colspan=3D"2" =
valign=3D"middle" style=3D"width: 112.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12.7px/normal Helvetica; min-height: 16px; "><SPAN =
class=3D"Apple-style-span" style=3D"line-height: =
19px;"><BR></SPAN></DIV></TD></TR></TBODY></TABLE><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV></BODY></HTML>=

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From sip-bounces@ietf.org Tue Jul 03 13:47:47 2007
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Subject: Re: [Sip] Security issues with ietf-sip-loop-fix
From: Scott Lawrence <slawrence@pingtel.com>
To: Cullen Jennings <fluffy@cisco.com>
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On Mon, 2007-07-02 at 19:35 -0700, Cullen Jennings wrote:

> Anyways, we need to figure out what to do next with this draft. There  
> are basically two paths forward that I see:
> 
> 1) reduce the scope so that it only deals with accidental loops and  
> provides no protection against malicious loops. I don't think this  
> would change any of the mechanisms in the draft but it would probably  
> change lots of the text around what security claims the draft made.
> 
> 2) try to extend or replace the mechanism with one that does protect  
> against malicious attack. A strong candidate to consider would be the  
> work in draft-sparks-sipping-max-breadth.
> 
> If we go forward with path 1, the value of this draft is probably not  
> super high because Max-Forwards deals with the the accidental loops  
> in most cases  (but not all cases). If we go with path 2, the WG will  
> need make sure it OK with the solution. I'm certainly open to other  
> paths forward if someone has a good idea.

While I agree that we need a stronger mechanism for the malicious case,
I have personally concluded that the earlier loop detection suggested by
ietf-sip-loop-fix is a good idea anyway.  Especially in those
configurations in which the maximum allowed value for Max-Forwards is
high (and I think the recommended default of 70 is _very_ high), it is
much better to detect the loop earlier than hop limit. 

I would be in favor of retaining some form of that recommendation.
Doing early loop detection will actually make the breadth-limiting
approach work better anyway, since it will allow non-looped branches to
proceed more quickly by pruning the loops sooner.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXecs project coordinator - SIPfoundry http://www.sipfoundry.org/sipXecs
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/



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> >>Chapter 9 says:
> >> 
> >>"in order for the mechanism to work, SBC-type-of-entities 
> >>must permit 
> >>DTLS, TLS, ICE, or HIP messages to be exchanged in the media path."
> >> 
> >>A small question for clarification: at what point must this 
> >>exchange 
> >>(two-way, I assume) work? As soon as the UAS has received 
> >>the INVITE?
> > 
> >Yes.
> 
> As we know from our RTPSEC discussions, one of the issues of using
> SBC-type-of-entities (or gates) is that you MAY NOT have end-to-end
> media plane connectivity as soon as the INVITE has been sent.

In such a situation, the authentication service will still be able 
to validate the signature.  This may be sufficient until or unless
replay attacks are causing havoc.  If It Ss Considered that replay
attacks are a small problem, then that's all you need to do and
breaking the media plane is fine.  However, as attackers get 
more sophisticated and capable of replay attacks, it will be 
necessary to use DTLS/TLS/HIP/ICE proof of possession of the 
private key.

One could also implement a system where edge SBCs perform
the DTLS/TLS/HIP/ICE validation on behalf of the endpoints;
this might be suitable for certain network (such as endpoints
that cannot or will not support DTLS/TLS/HIP/ICE themselves
but who need strong identity for phone calls for whitelists,
blacklists, and useful reptuation systems).

-d


> Regards,
> 
> Christer
> 
> 
>  
> >The DTLS, TLS, ICE, or HIP exchange would need to occur prior 
> >to displaying Caller-ID-like information to the user -- 
> >otherwise, an attacker could spoof that Caller-ID 
> >information.  It seems defective, to me, to display Caller-ID 
> >information that was spoofed to only later learn (after doing 
> >DTLS, TLS, ICE, or HIP) that the calling party had lied about 
> >their identity.  
> > 
> >There is some protection against such an attack that could 
> >reduce this risk (the Date: header is signed, so the attacker 
> >would need to obtain a relatively recent SIP request) -- that 
> >protection might be sufficient in networks that block 
> >exchange of packets prior to 200 Ok.  If an attacker was able 
> >to get ahold of such a SIP request, you would know it when 
> >the attacker failed the DTLS, TLS, ICE, or HIP exchange, 
> >which I suppose is better than nothing.
> > 
> > -d
> > 


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Subject: RE: [Sip] Certificate authentication in SIP
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Hi Jonathan,=20

The use case we are looking at is where the registrar is separate from
the edge proxy. While the edge proxy could validate the UA has access to
a cert issued by a trusted root, it doesn't necessarily mean the UA has
a valid subscription to network services. The edge proxy may/would not
have access to this type of registration data, thus the need to
authenticate from UA to registrar. The only use case I am aware of is
UAs authenticating directly to registrar's, for these same reasons.=20

Steve.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Wednesday, June 27, 2007 2:43 PM
To: Dean Willis
Cc: IETF SIP List; DRAGE,Keith (Keith)
Subject: Re: [Sip] Certificate authentication in SIP



Dean Willis wrote:

> Jonathan Rosenberg wrote:
>=20
>>Well, I'm going to be contrarian here. I'm not convinced that this is=20
>>needed.
>>
>>I think certificate based authentication is a great idea. However, I=20
>>am not sure I understand why TLS is not an appropriate solution.
>>
>=20
>=20
> I think it is very simple why TLS is not appropriate. TLS doesn't work

> across proxies, and would therefore require the edge proxy to do=20
> authentication.

So what? I think thats what ought to happen. I'd like to see some
specific use cases where this can't work with the edge proxy performing
the authentication. Keep in mind, we are talking about *certificate*
authentication; that doesn't (by definition) required any kind of
pre-arranged secret - only a common root CA.

-Jonathan R.
--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: RE: [Sip] Certificate authentication in SIP
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Thanks Vijay, and thanks Scott for the clarification.

The SIP cert auth requirements document currently lists a few use cases:

 - the certificate identifies a device
 - the certificate identifies a user

There could also be the case where the device certificate is mapped to a
user for subscription purposes, and there are probably others.

As Sumanth states, depending on the agreed upon requirements, the
solution could leave these types of specifics as out of scope and just
handle the transport and messaging between UA and registrar, or we could
go so far as to have certificate profiles and requirements and then work
those requirements with the appropriate groups.

Thanks.

Steve.=20

-----Original Message-----
From: Scott Lawrence [mailto:slawrence@pingtel.com]=20
Sent: Friday, June 29, 2007 12:32 PM
To: IETF SIP List; ietf-pkix@imc.org
Subject: Re: [Sip] Certificate authentication in SIP

On Fri, 2007-06-29 at 09:37 -0500, Vijay K. Gurbani wrote:
> Sumanth Channabasappa wrote:
> > And if we find that certificates need some work to support this=20
> > initiative (e.g., SIP identifiers as subjects), perhaps we can=20
> > present some of those requirements to other WGs. If we find an=20
> > existing solutions that can be used, good (and we can document them=20
> > as such :) ).
>=20
> Scott Lawrence and I have spent some time on this issue, i.e., SIP=20
> identifiers as subjects in X.509 certificates.  The latest version of=20
> the draft that includes pkix WG comments from Prague and the comments=20
> of the sip WG ADs and others was posted last week to the archives, and

> is available at
> http://tools.ietf.org/html/draft-gurbani-sip-domain-certs-05

One qualification - the draft above is limited to certificates as whose
subject is a SIP domain - not an individual.  The goal is to clarify how
such certificates are constructed and constrained, and how they should
be used to authenticate that a server is authoritative for a domain.

> Comments on this version would be extremely helpful.

--
Scott Lawrence  tel:+1-781-938-5306;ext=3D162 or =
sip:slawrence@pingtel.com
  sipXecs project coordinator - SIPfoundry
http://www.sipfoundry.org/sipXecs
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/



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Subject: RE: [Sip] Certificate authentication in SIP
Date: Tue, 3 Jul 2007 15:12:35 -0600
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From: "Steve Dotson" <S.Dotson@CableLabs.com>
To: "Sumanth Channabasappa" <sumanth@CableLabs.com>,
	"Peterson, Jon" <jon.peterson@neustar.biz>,
	"Dan Wing" <dwing@cisco.com>, "Jonathan Rosenberg" <jdrosen@cisco.com>, 
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
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Jon,
=20
I don't want to downplay the challenges of certificate management.
However, with TLS gaining traction in different forms, it would seem
these issues are being faced already. There are current PKI deployments
that could leverage existing investments, and certificates are starting
to be used and/or required in different ways in the SIP and mobile
space. Using large IPsec deployments as another example, some operators
are perfectly fine with managing IPsec relationships with shared
secrets, and some see the benefit of certificates.=20
=20
I honestly don't have much background on the "issues" with S/MIME or
AIB, and we very well may find after discussing the requirements that an
existing solution could be enhanced to meet the requirements. We are
obviously open and encourage re-use.
=20
Thanks.
=20
Steve.

________________________________

From: Sumanth Channabasappa [mailto:sumanth@cablelabs.com]=20
Sent: Friday, June 29, 2007 7:35 AM
To: Peterson, Jon; Dan Wing; Jonathan Rosenberg; DRAGE, Keith (Keith)
Cc: IETF SIP List
Subject: RE: [Sip] Certificate authentication in SIP


Jon,


But that's not really material - my note certainly wasn't an attempt to
rekindle or in any way glamorize S/MIME or AIB. It was rather a reminder
that no small part of the reason why those mechanisms didn't set the
world on fire was because of generic certificate enrollment problems,
user mobility requiring mobile certificate management, the
implausibility of certificates with SIP identifiers as subjects, and a
myriad of similar problems that have little to do with whether
authentication is end-to-end or what have you - they are rather rooted
in the easily-underestimated difficulty of managing the relationship
between SIP users, SIP devices, and certificates.
[S] Would this not be part of the reason that we want to tackle this
problem? There are large deployments which use certificates successfully
today (such as cable clients). There are potential deployments which
would like to use certificates for authentication in SIP networks,
without a specified explicit way to do this (or perhaps, lack the
understanding). Now, certificate management is a valid concern, and
potentially a hard problem, but that is only one part of the puzzle (and
may not apply to all deployments).=20
=20
I think we should start by identifying the requirements (since there
seems to be some confusion; and the I-D has not been discussed in the
WG) and see if existing or new solutions are required. And if we find
that certificates need some work to support this initiative (e.g., SIP
identifiers as subjects), perhaps we can present some of those
requirements to other WGs. If we find an existing solutions that can be
used, good (and we can document them as such :) ).=20

- s

=20


Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Friday, June 29, 2007 1:00 AM
> To: Peterson, Jon; 'Jonathan Rosenberg'; 'DRAGE, Keith (Keith)'
> Cc: 'IETF SIP List'
> Subject: RE: [Sip] Certificate authentication in SIP
>
>
> The certificate authentication would be used in place of today's
Digest
> authentication.
>
> S/MIME and AIB were never used where Digest is used; I don't see the
> relationship between what's on the table now and S/MIME and
> AIB -- except
> that they are two certificate-based authentication schemes,
> S/MIME and AIB
> are both intended to work end-to-end (between the two SIP
> peers desiring to
> establish communication with each other), whereas the certificate
> authentication being discussed is to replace ("enhance",
> whatever word you
> prefer) the username/password digest authentication.  Digest
> authentication
> isn't done between peers establishing communication with each
> other (except
> in a laboratory environment), but Digest is done to
> authenticate yourself to
> a SIP network so you can gain authorization to interact with that SIP
> network --- and that's what's on the table for certificate
> authentication.
>
> -d
>
>
> > -----Original Message-----
> > From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> > Sent: Wednesday, June 27, 2007 1:40 PM
> > To: Jonathan Rosenberg; DRAGE, Keith (Keith)
> > Cc: IETF SIP List
> > Subject: RE: [Sip] Certificate authentication in SIP
> >
> >
> > I also have to admit I'm a skeptical. Various forms of
> > non-hop-by-hop authentication with certificates were enabled
> > by S/MIME, especially in conjunction with entities like AIBs.
> > As far as I'm concerned, the mechanics have had their day in
> > court, and it didn't go well. We can grapple with the syntax
> > to try to find something slightly different that will
> > actually appeal to the implementation community, but I don't
> > think the problem was that we had the wrong syntax.
> >
> > Jon Peterson
> > NeuStar, Inc.
> >
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > > Sent: Tuesday, June 26, 2007 3:50 PM
> > > To: DRAGE, Keith (Keith)
> > > Cc: IETF SIP List
> > > Subject: Re: [Sip] Certificate authentication in SIP
> > >
> > >
> > > Well, I'm going to be contrarian here. I'm not convinced
> > that this is
> > > needed.
> > >
> > > I think certificate based authentication is a great idea.
> > > However, I am
> > > not sure I understand why TLS is not an appropriate solution.
> > >
> > > DRAGE, Keith (Keith) wrote:
> > >
> > > > (As WG chair)
> > > >
> > > >
> > > http://www.ietf.org/internet-drafts/draft-dotson-sip-certifica
> > > te-auth-03
> > > > .txt
> > > >
> > > > Describes a set of requirements for:
> > > >
> > > >    This document defines requirements for adding certificate
> > > >    authentication to the Session Initiation Protocol
> (SIP).  This
> > > >    document is being presented with the intention of
> > providing clear
> > > >    requirements to any potential solutions specifying
> certificate
> > > >    authentication within SIP networks.  Supporting certificate
> > > >    authentication in SIP would provide strong authentication and
> > > >    increase the types of possible deployment scenarios.
> > > >
> > > > (Before we go any further, please forget all about the solutions
> > > > document - that comes later and we are not dealing with it now)
> > > >
> > > > We need to decide whether there is support for a body of
> > > work in this
> > > > area, and therefore whether we should charter some
> > > requirements work in
> > > > the SIP WG.
> > > >
> > > > (Because this is security related we have agreed that
> SIP does the
> > > > requirements drafting and not SIPPING)
> > > >
> > > > So can I hear opinions of the WG on:
> > > >
> > > > -       whether this represents a problem space that
> the working group
> > > > should draft requirements on?
> > > >
> > > > -       whether the problem space exists but is
> something slightly
> > > > different, and if so what is that problem space?
> > > >
> > > > -       whether there is a more general problem that
> the security area
> > > > should be addressing, rather than the SIP group
> > addressing something
> > > > specific?
> > > >
> > > > -       based on your answers to the first three
> questions, whether this
> > > > draft is essentially in the right direction to be adopted
> > as the WG
> > > > draft assuming we create the charter item, or whether we
> > > need to seek
> > > > some other input draft?
> > > >
> > > > -       and finally, whether (assuming we go ahead with
> this work) there
> > > > is any work in any other IETF WG that we should take account of?
> > > >
> > > >
> > > > Regards
> > > >
> > > > Keith
> > > >
> > > >
> > > >
> > > > Regards
> > > >
> > > > Keith
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on
> current sip
> > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > >
> > >
> > > --
> > > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > > Cisco Fellow                                   Parsippany, NJ
> > > 07054-2711
> > > Cisco Systems
> > > jdrosen@cisco.com                              FAX: =20
> (973) 952-5050
> > > http://www.jdrosen.net <http://www.jdrosen.net/>
PHONE:
> (973) 952-5000
> > > http://www.cisco.com <http://www.cisco.com/>=20
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the
> application of sip
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>


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



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<HTML dir=3Dltr><HEAD><TITLE>RE: [Sip] Certificate authentication in =
SIP</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3132" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Jon,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I don't want to downplay the challenges of =
certificate=20
management. However, with TLS gaining traction in different forms, it =
would seem=20
these issues are being faced already. There are current PKI deployments =
that=20
could leverage existing investments, and certificates are starting to be =
used=20
and/or required in different ways in the SIP and mobile space. Using =
large IPsec=20
deployments as another example, some operators are perfectly fine with =
managing=20
IPsec relationships with shared secrets, and some see the benefit of=20
certificates. </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I honestly don't have much background on the =
"issues" with=20
S/MIME or AIB, and we very well may find after discussing the =
requirements that=20
an existing solution could be enhanced to meet the requirements. We are=20
obviously open and encourage re-use.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thanks.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D460180421-03072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Steve.</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> Sumanth Channabasappa=20
[mailto:sumanth@cablelabs.com] <BR><B>Sent:</B> Friday, June 29, 2007 =
7:35=20
AM<BR><B>To:</B> Peterson, Jon; Dan Wing; Jonathan Rosenberg; DRAGE, =
Keith=20
(Keith)<BR><B>Cc:</B> IETF SIP List<BR><B>Subject:</B> RE: [Sip] =
Certificate=20
authentication in SIP<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV id=3DidOWAReplyText24066 dir=3Dltr>
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 =
size=3D2>Jon,</FONT></DIV></DIV>
<DIV dir=3Dltr><BR><FONT size=3D2><BR>But that's not really material - =
my note=20
certainly wasn't an attempt to rekindle or in any way glamorize S/MIME =
or AIB.=20
It was rather a reminder that no small part of the reason why those =
mechanisms=20
didn't set the world on fire was because of generic certificate =
enrollment=20
problems, user mobility requiring mobile certificate management, the=20
implausibility of certificates with SIP identifiers as subjects, and a =
myriad of=20
similar problems that have little to do with whether authentication is=20
end-to-end or what have you - they are rather rooted in the=20
easily-underestimated difficulty of managing the relationship between =
SIP users,=20
SIP devices, and certificates.<BR>[S] Would this not be part of the =
reason that=20
we want to tackle this problem? There are large deployments which use=20
certificates successfully today (such as cable clients). There are =
potential=20
deployments which would like to use certificates for authentication in =
SIP=20
networks, without a specified explicit way to do this (or perhaps, lack =
the=20
understanding). Now, certificate management is a valid concern, and =
potentially=20
a hard problem, but that is only one part of the puzzle (and may not =
apply to=20
all deployments). </FONT></DIV>
<DIV dir=3Dltr><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2>I think we should start by identifying the =

requirements (since there seems to be some confusion; and the I-D has =
not been=20
discussed in the WG) and see if existing or new solutions are required. =
And if=20
we find that certificates need some work to support this initiative =
(e.g., SIP=20
identifiers as subjects), perhaps we can present some of those =
requirements to=20
other WGs. If we find an existing solutions that can be used, good (and =
we can=20
document them as such :) ). </FONT></DIV>
<P><FONT size=3D2>- s</FONT></P>
<P><FONT size=3D2>&nbsp;</P>
<DIV dir=3Dltr><BR>Jon Peterson<BR>NeuStar, Inc.<BR><BR>&gt; =
-----Original=20
Message-----<BR>&gt; From: Dan Wing [<A=20
href=3D"mailto:dwing@cisco.com">mailto:dwing@cisco.com</A>]<BR>&gt; =
Sent: Friday,=20
June 29, 2007 1:00 AM<BR>&gt; To: Peterson, Jon; 'Jonathan Rosenberg'; =
'DRAGE,=20
Keith (Keith)'<BR>&gt; Cc: 'IETF SIP List'<BR>&gt; Subject: RE: [Sip]=20
Certificate authentication in SIP<BR>&gt;<BR>&gt;<BR>&gt; The =
certificate=20
authentication would be used in place of today's Digest<BR>&gt;=20
authentication.<BR>&gt;<BR>&gt; S/MIME and AIB were never used where =
Digest is=20
used; I don't see the<BR>&gt; relationship between what's on the table =
now and=20
S/MIME and<BR>&gt; AIB -- except<BR>&gt; that they are two =
certificate-based=20
authentication schemes,<BR>&gt; S/MIME and AIB<BR>&gt; are both intended =
to work=20
end-to-end (between the two SIP<BR>&gt; peers desiring to<BR>&gt; =
establish=20
communication with each other), whereas the certificate<BR>&gt; =
authentication=20
being discussed is to replace ("enhance",<BR>&gt; whatever word =
you<BR>&gt;=20
prefer) the username/password digest authentication.&nbsp; =
Digest<BR>&gt;=20
authentication<BR>&gt; isn't done between peers establishing =
communication with=20
each<BR>&gt; other (except<BR>&gt; in a laboratory environment), but =
Digest is=20
done to<BR>&gt; authenticate yourself to<BR>&gt; a SIP network so you =
can gain=20
authorization to interact with that SIP<BR>&gt; network --- and that's =
what's on=20
the table for certificate<BR>&gt; authentication.<BR>&gt;<BR>&gt;=20
-d<BR>&gt;<BR>&gt;<BR>&gt; &gt; -----Original Message-----<BR>&gt; &gt; =
From:=20
Peterson, Jon [<A=20
href=3D"mailto:jon.peterson@neustar.biz">mailto:jon.peterson@neustar.biz<=
/A>]<BR>&gt;=20
&gt; Sent: Wednesday, June 27, 2007 1:40 PM<BR>&gt; &gt; To: Jonathan =
Rosenberg;=20
DRAGE, Keith (Keith)<BR>&gt; &gt; Cc: IETF SIP List<BR>&gt; &gt; =
Subject: RE:=20
[Sip] Certificate authentication in SIP<BR>&gt; &gt;<BR>&gt; =
&gt;<BR>&gt; &gt; I=20
also have to admit I'm a skeptical. Various forms of<BR>&gt; &gt; =
non-hop-by-hop=20
authentication with certificates were enabled<BR>&gt; &gt; by S/MIME, =
especially=20
in conjunction with entities like AIBs.<BR>&gt; &gt; As far as I'm =
concerned,=20
the mechanics have had their day in<BR>&gt; &gt; court, and it didn't go =
well.=20
We can grapple with the syntax<BR>&gt; &gt; to try to find something =
slightly=20
different that will<BR>&gt; &gt; actually appeal to the implementation=20
community, but I don't<BR>&gt; &gt; think the problem was that we had =
the wrong=20
syntax.<BR>&gt; &gt;<BR>&gt; &gt; Jon Peterson<BR>&gt; &gt; NeuStar,=20
Inc.<BR>&gt; &gt;<BR>&gt; &gt; &gt; -----Original Message-----<BR>&gt; =
&gt; &gt;=20
From: Jonathan Rosenberg [<A=20
href=3D"mailto:jdrosen@cisco.com">mailto:jdrosen@cisco.com</A>]<BR>&gt; =
&gt; &gt;=20
Sent: Tuesday, June 26, 2007 3:50 PM<BR>&gt; &gt; &gt; To: DRAGE, Keith=20
(Keith)<BR>&gt; &gt; &gt; Cc: IETF SIP List<BR>&gt; &gt; &gt; Subject: =
Re: [Sip]=20
Certificate authentication in SIP<BR>&gt; &gt; &gt;<BR>&gt; &gt; =
&gt;<BR>&gt;=20
&gt; &gt; Well, I'm going to be contrarian here. I'm not =
convinced<BR>&gt; &gt;=20
that this is<BR>&gt; &gt; &gt; needed.<BR>&gt; &gt; &gt;<BR>&gt; &gt; =
&gt; I=20
think certificate based authentication is a great idea.<BR>&gt; &gt; =
&gt;=20
However, I am<BR>&gt; &gt; &gt; not sure I understand why TLS is not an=20
appropriate solution.<BR>&gt; &gt; &gt;<BR>&gt; &gt; &gt; DRAGE, Keith =
(Keith)=20
wrote:<BR>&gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; (As WG chair)<BR>&gt; =
&gt; &gt;=20
&gt;<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; <A=20
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certifica">h=
ttp://www.ietf.org/internet-drafts/draft-dotson-sip-certifica</A><BR>&gt;=
=20
&gt; &gt; te-auth-03<BR>&gt; &gt; &gt; &gt; .txt<BR>&gt; &gt; &gt; =
&gt;<BR>&gt;=20
&gt; &gt; &gt; Describes a set of requirements for:<BR>&gt; &gt; &gt;=20
&gt;<BR>&gt; &gt; &gt; &gt;&nbsp;&nbsp;&nbsp; This document defines =
requirements=20
for adding certificate<BR>&gt; &gt; &gt; &gt;&nbsp;&nbsp;&nbsp; =
authentication=20
to the Session Initiation Protocol<BR>&gt; (SIP).&nbsp; This<BR>&gt; =
&gt; &gt;=20
&gt;&nbsp;&nbsp;&nbsp; document is being presented with the intention =
of<BR>&gt;=20
&gt; providing clear<BR>&gt; &gt; &gt; &gt;&nbsp;&nbsp;&nbsp; =
requirements to=20
any potential solutions specifying<BR>&gt; certificate<BR>&gt; &gt; &gt; =

&gt;&nbsp;&nbsp;&nbsp; authentication within SIP networks.&nbsp; =
Supporting=20
certificate<BR>&gt; &gt; &gt; &gt;&nbsp;&nbsp;&nbsp; authentication in =
SIP would=20
provide strong authentication and<BR>&gt; &gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;=20
increase the types of possible deployment scenarios.<BR>&gt; &gt; &gt;=20
&gt;<BR>&gt; &gt; &gt; &gt; (Before we go any further, please forget all =
about=20
the solutions<BR>&gt; &gt; &gt; &gt; document - that comes later and we =
are not=20
dealing with it now)<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; We =
need to=20
decide whether there is support for a body of<BR>&gt; &gt; &gt; work in=20
this<BR>&gt; &gt; &gt; &gt; area, and therefore whether we should =
charter=20
some<BR>&gt; &gt; &gt; requirements work in<BR>&gt; &gt; &gt; &gt; the =
SIP=20
WG.<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; (Because this is =
security=20
related we have agreed that<BR>&gt; SIP does the<BR>&gt; &gt; &gt; &gt;=20
requirements drafting and not SIPPING)<BR>&gt; &gt; &gt; &gt;<BR>&gt; =
&gt; &gt;=20
&gt; So can I hear opinions of the WG on:<BR>&gt; &gt; &gt; &gt;<BR>&gt; =
&gt;=20
&gt; &gt; -&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; whether this represents =
a=20
problem space that<BR>&gt; the working group<BR>&gt; &gt; &gt; &gt; =
should draft=20
requirements on?<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;=20
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; whether the problem space exists =
but=20
is<BR>&gt; something slightly<BR>&gt; &gt; &gt; &gt; different, and if =
so what=20
is that problem space?<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;=20
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; whether there is a more general =
problem=20
that<BR>&gt; the security area<BR>&gt; &gt; &gt; &gt; should be =
addressing,=20
rather than the SIP group<BR>&gt; &gt; addressing something<BR>&gt; &gt; =
&gt;=20
&gt; specific?<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;=20
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; based on your answers to the first =

three<BR>&gt; questions, whether this<BR>&gt; &gt; &gt; &gt; draft is=20
essentially in the right direction to be adopted<BR>&gt; &gt; as the =
WG<BR>&gt;=20
&gt; &gt; &gt; draft assuming we create the charter item, or whether =
we<BR>&gt;=20
&gt; &gt; need to seek<BR>&gt; &gt; &gt; &gt; some other input =
draft?<BR>&gt;=20
&gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; =
-&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and=20
finally, whether (assuming we go ahead with<BR>&gt; this work) =
there<BR>&gt;=20
&gt; &gt; &gt; is any work in any other IETF WG that we should take =
account=20
of?<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; =

Regards<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; Keith<BR>&gt; &gt; =
&gt;=20
&gt;<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; =
&gt;=20
Regards<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt; Keith<BR>&gt; &gt; =
&gt;=20
&gt;<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt; &gt; &gt;=20
_______________________________________________<BR>&gt; &gt; &gt; &gt; =
Sip=20
mailing list&nbsp; <A=20
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>&gt;=20
&gt; &gt; &gt; This list is for NEW development of the core SIP =
Protocol<BR>&gt;=20
&gt; &gt; &gt; Use sip-implementors@cs.columbia.edu for questions =
on<BR>&gt;=20
current sip<BR>&gt; &gt; &gt; &gt; Use sipping@ietf.org for new =
developments on=20
the<BR>&gt; &gt; application of sip<BR>&gt; &gt; &gt; &gt;<BR>&gt; &gt;=20
&gt;<BR>&gt; &gt; &gt; --<BR>&gt; &gt; &gt; Jonathan D. Rosenberg,=20
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
600 Lanidex Plaza<BR>&gt; &gt; &gt; Cisco=20
Fellow&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&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;=20
Parsippany, NJ<BR>&gt; &gt; &gt; 07054-2711<BR>&gt; &gt; &gt; Cisco=20
Systems<BR>&gt; &gt; &gt;=20
jdrosen@cisco.com&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;&nbsp;&nbsp;&nbsp;&nbsp;=20
FAX:&nbsp;&nbsp;<BR>&gt; (973) 952-5050<BR>&gt; &gt; &gt; <A=20
href=3D"http://www.jdrosen.net/">http://www.jdrosen.net</A>&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
PHONE:<BR>&gt; (973) 952-5000<BR>&gt; &gt; &gt; <A=20
href=3D"http://www.cisco.com/">http://www.cisco.com</A><BR>&gt; &gt; =
&gt;<BR>&gt;=20
&gt; &gt;<BR>&gt; &gt; &gt;=20
_______________________________________________<BR>&gt; &gt; &gt; Sip =
mailing=20
list&nbsp; <A=20
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>&gt;=20
&gt; &gt; This list is for NEW development of the core SIP =
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sip<BR>&gt;=20
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list&nbsp; <A=20
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&gt; This list is for NEW development of the core SIP Protocol<BR>&gt; =
&gt; Use=20
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href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
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list is for NEW development of the core SIP Protocol<BR>Use=20
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sipping@ietf.org for new developments on the application of=20
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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Dan Wing" <dwing@cisco.com>
To: "'Steve Dotson'" <S.Dotson@CableLabs.com>
Subject: RE: [Sip] Certificate authentication in SIP
Date: Tue, 3 Jul 2007 14:34:22 -0700
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(removed some direct CC's.) 

> -----Original Message-----
> From: Steve Dotson [mailto:S.Dotson@CableLabs.com] 
> Sent: Tuesday, July 03, 2007 2:13 PM
> To: Sumanth Channabasappa; Peterson, Jon; Dan Wing; Jonathan 
> Rosenberg; DRAGE, Keith (Keith)
> Cc: IETF SIP List
> Subject: RE: [Sip] Certificate authentication in SIP
> 
> Jon,
>  
> I don't want to downplay the challenges of certificate 
> management. However, with TLS gaining traction in different 
> forms, it would seem these issues are being faced already. 
> There are current PKI deployments that could leverage 
> existing investments, and certificates are starting to be 
> used and/or required in different ways in the SIP and mobile 
> space. Using large IPsec deployments as another example, some 
> operators are perfectly fine with managing IPsec 
> relationships with shared secrets, and some see the benefit 
> of certificates. 
>  
> I honestly don't have much background on the "issues" with 
> S/MIME or AIB, and we very well may find after discussing the 
> requirements that an existing solution could be enhanced to 
> meet the requirements. We are obviously open and encourage re-use.

AIB copies the headers into a separate MIME part and uses S/MIME to sign
that part.  The unfortunate lack of MIME support in SIP is a significant
barrier to the successful deployment of anything using multipart MIME
messages, such as AIB.  From the last SIPit,
http://www.sipforum.org/content/view/287/171/: 

    >
    >    2%  I break if someone sends me multipart/mime
    >   24%  I pretend multipart/mime doesn't exist if someone
    >        sends it to me
    >   24%  I ignore multipart/mime but will proxy it or hand
    >        it to my application if it shows up
    >   10%  I try to do something useful with multipart/mime I
    >        receive, but I never send it
    >    4%  I ignore multipart/mime that I receive, but I try
    >        to do something useful with multipart/mime I send
    >   24%  I try to do something useful with multipart/mime I
    >        send and receive
    >   12%  Other
    >

If you re-used AIB just for Register, the situation may be better, as you
only need the SIP proxy to ignore the multipart portion of the message.

You might also read draft-camarillo-sip-body-handling-01.txt on this topic.

-d


> Thanks.
>  
> Steve.
> 
> ________________________________
> 
> From: Sumanth Channabasappa [mailto:sumanth@cablelabs.com] 
> Sent: Friday, June 29, 2007 7:35 AM
> To: Peterson, Jon; Dan Wing; Jonathan Rosenberg; DRAGE, Keith (Keith)
> Cc: IETF SIP List
> Subject: RE: [Sip] Certificate authentication in SIP
> 
> 
> Jon,
> 
> 
> But that's not really material - my note certainly wasn't an 
> attempt to rekindle or in any way glamorize S/MIME or AIB. It 
> was rather a reminder that no small part of the reason why 
> those mechanisms didn't set the world on fire was because of 
> generic certificate enrollment problems, user mobility 
> requiring mobile certificate management, the implausibility 
> of certificates with SIP identifiers as subjects, and a 
> myriad of similar problems that have little to do with 
> whether authentication is end-to-end or what have you - they 
> are rather rooted in the easily-underestimated difficulty of 
> managing the relationship between SIP users, SIP devices, and 
> certificates.
> [S] Would this not be part of the reason that we want to 
> tackle this problem? There are large deployments which use 
> certificates successfully today (such as cable clients). 
> There are potential deployments which would like to use 
> certificates for authentication in SIP networks, without a 
> specified explicit way to do this (or perhaps, lack the 
> understanding). Now, certificate management is a valid 
> concern, and potentially a hard problem, but that is only one 
> part of the puzzle (and may not apply to all deployments). 
>  
> I think we should start by identifying the requirements 
> (since there seems to be some confusion; and the I-D has not 
> been discussed in the WG) and see if existing or new 
> solutions are required. And if we find that certificates need 
> some work to support this initiative (e.g., SIP identifiers 
> as subjects), perhaps we can present some of those 
> requirements to other WGs. If we find an existing solutions 
> that can be used, good (and we can document them as such :) ). 
> 
> - s
> 
>  
> 
> 
> Jon Peterson
> NeuStar, Inc.
> 
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com]
> > Sent: Friday, June 29, 2007 1:00 AM
> > To: Peterson, Jon; 'Jonathan Rosenberg'; 'DRAGE, Keith (Keith)'
> > Cc: 'IETF SIP List'
> > Subject: RE: [Sip] Certificate authentication in SIP
> >
> >
> > The certificate authentication would be used in place of 
> today's Digest
> > authentication.
> >
> > S/MIME and AIB were never used where Digest is used; I don't see the
> > relationship between what's on the table now and S/MIME and
> > AIB -- except
> > that they are two certificate-based authentication schemes,
> > S/MIME and AIB
> > are both intended to work end-to-end (between the two SIP
> > peers desiring to
> > establish communication with each other), whereas the certificate
> > authentication being discussed is to replace ("enhance",
> > whatever word you
> > prefer) the username/password digest authentication.  Digest
> > authentication
> > isn't done between peers establishing communication with each
> > other (except
> > in a laboratory environment), but Digest is done to
> > authenticate yourself to
> > a SIP network so you can gain authorization to interact 
> with that SIP
> > network --- and that's what's on the table for certificate
> > authentication.
> >
> > -d
> >
> >
> > > -----Original Message-----
> > > From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> > > Sent: Wednesday, June 27, 2007 1:40 PM
> > > To: Jonathan Rosenberg; DRAGE, Keith (Keith)
> > > Cc: IETF SIP List
> > > Subject: RE: [Sip] Certificate authentication in SIP
> > >
> > >
> > > I also have to admit I'm a skeptical. Various forms of
> > > non-hop-by-hop authentication with certificates were enabled
> > > by S/MIME, especially in conjunction with entities like AIBs.
> > > As far as I'm concerned, the mechanics have had their day in
> > > court, and it didn't go well. We can grapple with the syntax
> > > to try to find something slightly different that will
> > > actually appeal to the implementation community, but I don't
> > > think the problem was that we had the wrong syntax.
> > >
> > > Jon Peterson
> > > NeuStar, Inc.
> > >
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > > > Sent: Tuesday, June 26, 2007 3:50 PM
> > > > To: DRAGE, Keith (Keith)
> > > > Cc: IETF SIP List
> > > > Subject: Re: [Sip] Certificate authentication in SIP
> > > >
> > > >
> > > > Well, I'm going to be contrarian here. I'm not convinced
> > > that this is
> > > > needed.
> > > >
> > > > I think certificate based authentication is a great idea.
> > > > However, I am
> > > > not sure I understand why TLS is not an appropriate solution.
> > > >
> > > > DRAGE, Keith (Keith) wrote:
> > > >
> > > > > (As WG chair)
> > > > >
> > > > >
> > > > http://www.ietf.org/internet-drafts/draft-dotson-sip-certifica
> > > > te-auth-03
> > > > > .txt
> > > > >
> > > > > Describes a set of requirements for:
> > > > >
> > > > >    This document defines requirements for adding certificate
> > > > >    authentication to the Session Initiation Protocol
> > (SIP).  This
> > > > >    document is being presented with the intention of
> > > providing clear
> > > > >    requirements to any potential solutions specifying
> > certificate
> > > > >    authentication within SIP networks.  Supporting certificate
> > > > >    authentication in SIP would provide strong 
> authentication and
> > > > >    increase the types of possible deployment scenarios.
> > > > >
> > > > > (Before we go any further, please forget all about 
> the solutions
> > > > > document - that comes later and we are not dealing 
> with it now)
> > > > >
> > > > > We need to decide whether there is support for a body of
> > > > work in this
> > > > > area, and therefore whether we should charter some
> > > > requirements work in
> > > > > the SIP WG.
> > > > >
> > > > > (Because this is security related we have agreed that
> > SIP does the
> > > > > requirements drafting and not SIPPING)
> > > > >
> > > > > So can I hear opinions of the WG on:
> > > > >
> > > > > -       whether this represents a problem space that
> > the working group
> > > > > should draft requirements on?
> > > > >
> > > > > -       whether the problem space exists but is
> > something slightly
> > > > > different, and if so what is that problem space?
> > > > >
> > > > > -       whether there is a more general problem that
> > the security area
> > > > > should be addressing, rather than the SIP group
> > > addressing something
> > > > > specific?
> > > > >
> > > > > -       based on your answers to the first three
> > questions, whether this
> > > > > draft is essentially in the right direction to be adopted
> > > as the WG
> > > > > draft assuming we create the charter item, or whether we
> > > > need to seek
> > > > > some other input draft?
> > > > >
> > > > > -       and finally, whether (assuming we go ahead with
> > this work) there
> > > > > is any work in any other IETF WG that we should take 
> account of?
> > > > >
> > > > >
> > > > > Regards
> > > > >
> > > > > Keith
> > > > >
> > > > >
> > > > >
> > > > > Regards
> > > > >
> > > > > Keith
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sipping@ietf.org for new developments on the
> > > application of sip
> > > > >
> > > >
> > > > --
> > > > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > > > Cisco Fellow                                   Parsippany, NJ
> > > > 07054-2711
> > > > Cisco Systems
> > > > jdrosen@cisco.com                              FAX:  
> > (973) 952-5050
> > > > http://www.jdrosen.net <http://www.jdrosen.net/>        
>                   PHONE:
> > (973) 952-5000
> > > > http://www.cisco.com <http://www.cisco.com/> 
> > > >
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> >
> 
> 
> _______________________________________________
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inline.

Dean Willis wrote:

>> So what? I think thats what ought to happen. I'd like to see some 
>> specific use cases where this can't work with the edge proxy 
>> performing the authentication. Keep in mind, we are talking about 
>> *certificate* authentication; that doesn't (by definition) required 
>> any kind of pre-arranged secret - only a common root CA.
> 
> My edge proxy might be provided by MCI, but I might be using a chat 
> service from another provider overseas. This other provider might be 
> willing to trust the certificate for authentication (given that I have a 
> signed cert from MCI, who in turn has a signed CA cert from a recognized 
> op-level CA). However, the chat provider might not be willing to just 
> accept P-Asserted-Identity from MCI as an authentication mechanism, 
> since said provider does not have a trusted peering arrangement with MCI 
> and therefore ANYBODY could spoof my P-Asserted-Identity.

In a case where your chat provider has no relationship whatsoever with 
this edge proxy, for what purpose would you be connecting to it in the 
first place? You should connect directly to your chat provider offering 
the service. Then, if you have a cert - great - authenticate to it using 
mutual TLS.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Steve Dotson wrote:
> Hi Jonathan, 
> 
> The use case we are looking at is where the registrar is separate from
> the edge proxy. 

Sure. I think that is a valid case.

> While the edge proxy could validate the UA has access to
> a cert issued by a trusted root, it doesn't necessarily mean the UA has
> a valid subscription to network services.

No. But, such a proxy could assert the identity it verified.

> The edge proxy may/would not
> have access to this type of registration data, thus the need to
> authenticate from UA to registrar. 

You lost me here. With mutual TLS, the client can authenticate itself to 
the edge proxy. The edge proxy can assert the authenticated identity 
towards the registrar, ala P-A-ID or even better, SIP Identity. THe 
registrar then authorizes access, and rejects the request if the user is 
not authorized.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Christer,

My concern with P-Called-Party-ID is similar to my concern on 
History-Info. On History-Info, the draft states:

    Firstly, it would cause the Request-URI to be relegated to nothing
    more than an indicator of the next hop for the request, identical
    exactly to the purpose of the Route header field.  This results in
    two things in the SIP specification which do exactly the same thing.
    Worse still, this is not just for some small feature of SIP (where
    such duplication might not be a big deal), but rather, it would be a
    duplication of SIP's primary function - routing of a call towards a
    target.

This concern applies exactly to P-Called-Party-ID. Indeed, its even 
worse, since now we would have, in a request:

Request-URI
To
P-Called-Party-ID
History-Info
Route

and I challenge you to explain to implementors what the differences are. 
What would happen when they are inconsistent?

My objective was to reduce this count. Indeed I was hopeful that 
UA-loose-route would allow us to deprecate P-Called-Party-ID, not the 
other way around. WIth UA-loose-route, we end up with:

Request-URI: identifies the target
Route: tells you how to get to the target
To: tells you the original target
History info: sequence of identities of targets

which is much cleaner.

So, functionally, P-Called-Party-ID could work. Any number of things 
*can* work.

-Jonathan R.


Christer Holmberg (JO/LMF) wrote:
> Hi,
> 
>> It is entirely appropriate at this stage to put forward 
>> alternatives. It is even more appropriate to do it now rather than
>> in 6 months time.
>> 
>> However please do not expect the editor to elaborate them. If the
>> WG has ideas, send text, either by individual author drafts, or by
>> the mailing list.
> 
> Regarding the P-Called-Party header, there already IS an RFC
> describing the header, and what it is used for. So, IF it already
> solves the problem (RFC3455 does say that the header is used to
> convey the AOR to the UAS), why would we need a separate draft? The
> only additional information that a draft could contain are the
> use-cases where receiving the AOR is important - but those use-cases
> are solution independent and already described in Jonathan's draft.
> 
> Of course, if we want to define a NEW header, or a NEW URI parameter,
> that may be done in a separate draft.
> 
> But, no matter how we do it, I think the author should at least
> indicate whether he has already thought about the alternatives that
> have been presented, and why he thinks that they don't work.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
>>> -----Original Message----- From: Christer Holmberg (JO/LMF) 
>>> [mailto:christer.holmberg@ericsson.com] Sent: Thursday, June 14,
>>> 2007 10:51 AM To: DRAGE, Keith (Keith); Jonathan Rosenberg; IETF
>>> SIP List Subject: RE: [Sip] UA Loose Routing
>>> 
>>> 
>>> Hi Keith,
>>> 
>>> The text you sent says:
>>> 
>>> "Consensus noted that we do need some sort of solution for
>> the general
>>> problem of delivery of URI parameters across contact-routing 
>>> operations."
>>> 
>>> I have no problems with that.
>>> 
>>> I also have no problem with using Jonathan's draft as base line,
>>>  because the draft discusses the problem in a good way.
>>> 
>>> But, still: as the text says, the agreement was to come up
>> with SOME
>>> SORT OF SOLUTION, which to me does not mean that we have agreed
>>> to adopt the specific solution in Jonathan's draft. We should
>>> also be "allowed" to discuss alternative solutions, whether they
>>> 
>> are based on
>>> P-Called-Party, some other header, some URI parameter, or
>> whatever...
>>> Maybe those alternative solutions do not work, and then we should
>>>  describe that (as Jonathan has done for the To header- and
>>> History header alternatives).
>>> 
>>> Regards,
>>> 
>>> Christer
>>> 
>>> 
>>> 
>>> 
>>>> -----Original Message----- From: DRAGE, Keith (Keith)
>>>> [mailto:drage@alcatel-lucent.com] Sent: 14. kesäkuuta 2007
>>>> 12:40 To: Christer Holmberg (JO/LMF); Jonathan Rosenberg; IETF
>>>> SIP List Subject: RE: [Sip] UA Loose Routing
>>>> 
>>>> You appear to be questioning whether the solution was
>> adopted. The
>>>> minutes state:
>>>> 
>>>> 
>> -------------------------------------------------------------------
>> 
>>>> Topic: GRUU and Supporting Drafts Led by Jonathan Rosenberg 
>>>> Slides included in proceedings Read:
>>>> draft-ietf-sip-gruu-11,draft-rosenberg-sip-ua-loose-route-00
>>>> 
>>>> Change since last version reviewed.
>>>> 
>>>> Issue: GRUU subtype names. The room demonstrated a slight
>>> preference
>>>> for the names used in -11 over known alternatives, but
>> nobody seems
>>>> particularly enamored with these names and the floor
>>> remains open to
>>>> suggestions.
>>>> 
>>>> Discussion on ua-loose-route draft deferred to next session.
>>>> 
>>>> 
>>>> Session 2: Friday 0900-1130 Grand Ballroom C
>>>> 
>>>> 
>>>> Topic: UA-loose-route, continued from previous session.
>>>> 
>>>> Consensus noted that we do need some sort of solution for
>>> the general
>>>> problem of delivery of URI parameters across contact-routing 
>>>> operations.
>>>> 
>>>> Issue: Concerns raised about backward compatibility. Noted that
>>>>  compliant P-CSCF should be OK.  Suggested that we have a
>>> design team
>>>> revisit call-flows and analyze for backward compatibility
>>>> issues.
>>>> 
>>>> Issue: Does GRUU normatively depend on this work? Resolved: 
>>>> GRUU does not reference this document. However, some
>>> participants feel
>>>> it should, as the benefit of GRUU cannot be fully
>> recognized in the
>>>> absence of this specification.  The chairs asked for a
>> hum, and the
>>>> consensus of the room is that GRUU and ua-loose-route can
>>>> proceed independently.
>>>> 
>>>> Question: Adopt this draft as baseline text for a WG
>> effort, should
>>>> the ADs approve the milestone? Consensus to adopt noted by
>>>> chairs. To-do: Chairs to work with AD to set new milestone.
>>>> 
>> ----------------------------------------------------------------------
>> 
>>>> The minutes quite clearly say "Adopt this draft as baseline
>>> text for a
>>>> WG effort" and given that the charter item now exists, then
>>> I believe
>>>> that we are entitled to call for the editor (Jonathan) to
>>> submit the
>>>> current text as draft-ietf-sip-ua-loose-route.
>>>> 
>>>> I agree the file name says loose-route, but I don'y think
>> we should
>>>> change that. After all, we lose that filename when it gets
>>> published
>>>> as an RFC anyway.
>>>> 
>>>> The charter item states
>>>> 
>>>> Dec 2007    Delivering request-URI and parameters to UAS via 
>>>> proxy to WGLC Feb 2008    Delivering request-URI and parameters
>>>> to UAS via proxy to IESG (PS)
>>>> 
>>>> I assume that you are OK with this.
>>>> 
>>>> What I would suggest is that you address specific comments to
>>>> the title of the deliverable:
>>>> 
>>>> Applying Loose Routing to Session Initiation Protocol (SIP)
>>>> User Agents (UA)
>>>> 
>>>> And to the abstract:
>>>> 
>>>> Abstract
>>>> 
>>>> A key part of the behavior of the Session Initiation
>>> Protocol (SIP)
>>>> is that SIP proxies rewrite the Request-URI as a request moves 
>>>> throughout the network.  Over the years, experience has
>>> shown this
>>>> to be problematic.  It makes it difficult to use Request URI
>>>> for service invocation, complicates emergency services, makes
>>>> it
>>> more complex
>>>> to support aliases, and so on.  Architecturally, it confounds
>>>> the concepts of address and route.  This document proposes
>> to change
>>>> this through a new mechanism called UA loose routing.
>>>> 
>>>> As you think appropriate to your concerns.
>>>> 
>>>> Alternatively, if you (or indeed any other SIP WG member)
>> do have a
>>>> completely different solution, then there is still time
>> to post an
>>>> individual draft or post a skeleton of a mechanism to the list.
>>>> 
>>>> 
>>>> Regards
>>>> 
>>>> Keith
>>>> 
>>>>> -----Original Message----- From: Christer Holmberg (JO/LMF) 
>>>>> [mailto:christer.holmberg@ericsson.com] Sent: Wednesday, June
>>>>> 13, 2007 8:52 PM To: DRAGE, Keith (Keith); Jonathan
>>>>> Rosenberg; IETF SIP List Subject: RE: [Sip] UA Loose Routing
>>>>> 
>>>>> 
>>>>> Hi,
>>>>> 
>>>>> We can call it loose route if we want, but I think the
>>>> initial focus
>>>>> should be the problem we are going to solve (I don't think
>>>> we agreed
>>>>> on a specific solution at #67), and we should look into
>>> alternative
>>>>> solutions (the draft does that in some extent, but
>> other possible
>>>>> alternatives have also been mentioned).
>>>>> 
>>>>> Or, do you prefer to have separate drafts describing
>> alternative
>>>>> solutions for the problem?
>>>>> 
>>>>> Regards,
>>>>> 
>>>>> Christer
>>>>> 
>>>>> 
>>>>> 
>>>>>> -----Original Message----- From: DRAGE, Keith (Keith)
>>>>>> [mailto:drage@alcatel-lucent.com] Sent: 13. kesäkuuta 2007
>>>>>> 22:25 To: Jonathan Rosenberg; IETF SIP List Subject: RE:
>>>>>> [Sip] UA Loose Routing
>>>>>> 
>>>>>> (As WG chair)
>>>>>> 
>>>>>> As well as agreeing to charter the item, we actually agreed
>>>>>> 
>>>>> at IETF#67
>>>>>> that this draft was a suitable candidate to fulfil that
>>>>> charter item.
>>>>>> Therefore unless issues are raised with this, I would
>>> request the
>>>>>> editor (as Jonathan is editor) to submit the next
>>> version of this
>>>>>> document as draft-ietf-sip-ua-loose-route-00.
>>>>>> 
>>>>>> Regards
>>>>>> 
>>>>>> Keith
>>>>>> 
>>>>>>> -----Original Message----- From: Jonathan Rosenberg
>>>>>>> [mailto:jdrosen@cisco.com] Sent: Wednesday, June 13, 2007
>>>>>>> 4:02 AM To: IETF SIP List Subject: [Sip] UA Loose Routing
>>>>>>> Issue #1: Backwards
>>>> compatibility
>>>>>>> I've recently revised my draft on UA loose routing. You
>>>>>>> can
>>>>>> pick it up
>>>>>>> here:
>>>>>>> 
>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
>>>>>>> se-route-01.txt
>>>>>>> 
>>>>>>> This draft is meant to fulfill one of our charter items: 
>>>>>>> Feb 2008	  	Delivering request-URI and parameters to UAS
>>>>>>> via proxy to IESG (PS)
>>>>>>> 
>>>>>>> As folks may recall, the GRUU spec used to have these
>>>>> things called
>>>>>>> grids, which allowed a UA to add its own parameters to
>>>>>>> the
>>>>>> GRUU before
>>>>>>> handing it out. Several GRUU revisions back, this
>>>>>>> function
>>>>>> got yanked
>>>>>>> because it was recognized that the problem was more
>>>>>>> general
>>>>>> than GRUU,
>>>>>>> and would be useful for AOR as well.
>>>>>>> 
>>>>>>> Two IETFs back I presented this draft briefly during a
>>>>> SIP session.
>>>>>>> There was a lot of interest and people liked it
>>>> architecturally,
>>>>>>> however there were two open issues that needed to get
>>>>>> resolved. This
>>>>>>> mail is meant to start discussion on one of them -
>> backwards
>>>>>>> compatibility.
>>>>>>> 
>>>>>>> THe mechanism basically changes the way proxies handle
>>>> requests.
>>>>>>> Instead of rewriting the request URI for out-of-dialog
>>>> requests,
>>>>>>> they'll push a route header and leave the r-uri alone.
>>>>>>> This
>>>>>> allows the
>>>>>>> original R-URI and any parameters it contains to get
>>>>>> delivered to the
>>>>>>> UAS.
>>>>>>> 
>>>>>>> So, the main issue is, what kinds of backwards
>>>>>> compatibility problems
>>>>>>> will this cause. The draft provides a basic mechanism
>>>>> whereby a UA
>>>>>>> includes an option tag in the REGISTER, and the registrar
>>>>>>> 
>>>>> indicates
>>>>>>> that it is using this mechanism or not. So we can
>>>> negotiate this
>>>>>>> between a UA and registrar easily. Now, the problem is
>>>>>>> with intermediate proxies. If there is an edge proxy
>>>>>>> between the
>>>>>> UA and its
>>>>>>> registrar, it will now see requests targeted to the UA,
>>>>>> which contain
>>>>>>> a Route header beyond itself, even though it thinks its
>>>>>>> the
>>>>>> edge proxy
>>>>>>> and the last point of contact before the UA. Technically,
>>>>>>> 
>>>>>> if this edge
>>>>>>> proxy is compliant to rfc3261 it would work. However, the
>>>>>>> 
>>>>>> question is
>>>>>>> whether proxies in the wild would properly work in
>> this case.
>>>>>>> If folks can comment based on their own products, or
>>>>>>> their experiences, on whether they think this would work,
>>>>>>> or
>>>>>> concerns they
>>>>>>> have, that would be great. THere are solutions that would
>>>>>>> 
>>>>>> allow us to
>>>>>>> disable the mechanism when intermediate proxies don't
>>>>>> support it, but
>>>>>>> they are more complex and I'd rather avoid it.
>>>>>>> 
>>>>>>> THanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D.
>>>>>>> 600
>>> Lanidex Plaza
>>>>>>> Cisco Fellow
>> Parsippany, NJ
>>>>>>> 07054-2711 Cisco Systems jdrosen@cisco.com
>>>>>>> FAX:
>>>>> (973) 952-5050
>>>>>>> http://www.jdrosen.net                         PHONE:
>>>>> (973) 952-5000
>>>>>>> http://www.cisco.com
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________ Sip
>>>>>>> mailing list  https://www1.ietf.org/mailman/listinfo/sip 
>>>>>>> This list is for NEW development of the core SIP
>> Protocol Use
>>>>>>> sip-implementors@cs.columbia.edu for questions on
>>>> current sip Use
>>>>>>> sipping@ietf.org for new developments on the
>>> application of sip
>>>>>> 
>>>>>> _______________________________________________ Sip mailing
>>>>>> list  https://www1.ietf.org/mailman/listinfo/sip This list
>>>>>> is for NEW development of the core SIP Protocol Use 
>>>>>> sip-implementors@cs.columbia.edu for questions on
>>> current sip Use
>>>>>> sipping@ietf.org for new developments on the
>> application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] UA Loose Routing Issue #1: Backwards compatibility
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Sorry, the draft is not clear on this. It also works in cases where the 
proxies translation logic is NOT based on registrations.

The basic idea is to apply the ;lr param in this case. So if the proxy 
looks up the request URI and find that the result is a SIP URI with the 
;lr param, it pushes that into the route set rather than rewriting the 
R-URI. Without the ;lr param, it rewrites the R-URI as it does today.
This is merely applying the exactly same logic we have for mid-dialog 
signaling, but for the initial dialog-forming request. I think there is 
some elegance in that.

So, if we consider this MGC case, the routing logic in the proxy that 
points to these old MGC would have entries which omit the ;lr parameter, 
and those pointing to ua-loose-route compliant ones would include it.

-Jonathan R.

Christer Holmberg (JO/LMF) wrote:
> Hi,
> 
> At least with existing MGCs you will have backward compability issues.
> 
> MGCs normally route SIP-to-ISUP calls based on the Request-URI, and if the Request-URI contains e.g. a sos servicetype URN, or a 800 service number, the MGC will not be able to route on those.
> 
> Regards,
> 
> Christer
> 
> 
> 
>  
> 
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
>> Sent: 13. kesäkuuta 2007 6:02
>> To: IETF SIP List
>> Subject: [Sip] UA Loose Routing Issue #1: Backwards compatibility
>>
>> I've recently revised my draft on UA loose routing. You can 
>> pick it up here:
>> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
>> se-route-01.txt
>>
>> This draft is meant to fulfill one of our charter items:
>> Feb 2008	  	Delivering request-URI and parameters 
>> to UAS via proxy to 
>> IESG (PS)
>>
>> As folks may recall, the GRUU spec used to have these things 
>> called grids, which allowed a UA to add its own parameters to 
>> the GRUU before handing it out. Several GRUU revisions back, 
>> this function got yanked because it was recognized that the 
>> problem was more general than GRUU, and would be useful for 
>> AOR as well.
>>
>> Two IETFs back I presented this draft briefly during a SIP session. 
>> There was a lot of interest and people liked it 
>> architecturally, however there were two open issues that 
>> needed to get resolved. This mail is meant to start 
>> discussion on one of them - backwards compatibility.
>>
>> THe mechanism basically changes the way proxies handle 
>> requests. Instead of rewriting the request URI for 
>> out-of-dialog requests, they'll push a route header and leave 
>> the r-uri alone. This allows the original R-URI and any 
>> parameters it contains to get delivered to the UAS.
>>
>> So, the main issue is, what kinds of backwards compatibility 
>> problems will this cause. The draft provides a basic 
>> mechanism whereby a UA includes an option tag in the 
>> REGISTER, and the registrar indicates that it is using this 
>> mechanism or not. So we can negotiate this between a UA and 
>> registrar easily. Now, the problem is with intermediate 
>> proxies. If there is an edge proxy between the UA and its 
>> registrar, it will now see requests targeted to the UA, which 
>> contain a Route header beyond itself, even though it thinks 
>> its the edge proxy and the last point of contact before the 
>> UA. Technically, if this edge proxy is compliant to rfc3261 
>> it would work. However, the question is whether proxies in 
>> the wild would properly work in this case.
>>
>> If folks can comment based on their own products, or their 
>> experiences, on whether they think this would work, or 
>> concerns they have, that would be great. THere are solutions 
>> that would allow us to disable the mechanism when 
>> intermediate proxies don't support it, but they are more 
>> complex and I'd rather avoid it.
>>
>> THanks,
>> Jonathan R.
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Cisco Fellow                                   Parsippany, NJ 
>> 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.com
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip 
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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

Salvatore Loreto wrote:
> Hi Jonathan,
> 
> about the third redirection meaning:
> 
>    3.  The server is asking the client to route through an additional
>        proxy prior to visiting it, so that the recursed request
>        generated by the client pushes an additional Route onto the Route
>        set.
> 
> I was wondering if it should be possible insert more than only one contact 
> into the existing route set.

No.

> I'd like to know your opinion, because in my understanding it should not possible 
> due to the "alternative" meaning of different locations into the the Contact 
> header field; in fact the RFC3261 states:
> 
>    "When a redirect server returns a 3xx response to a request, it
>    populates the list of (one or more) alternative locations into the
>    Contact header field.  An "expires" parameter to the Contact header
>    field values may also be supplied to indicate the lifetime of the
>    Contact data."
> 
> 
> About the persistence of the redirection for a mid-dialog request,
> I am not sure it should persist just for the duration of the transaction, 
> instead the persistence should be managed by the "expires" parameter.

The problem is - under what conditions would the proxy utilize this? For 
all requests whose destination would otherwise be the redirecting UA? 
Only ones whose R-URI also matches?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Dale.Worley@comcast.net wrote:
>    From: Jonathan Rosenberg <jdrosen@cisco.com>
> 
>    THe mechanism basically changes the way proxies handle requests. Instead 
>    of rewriting the request URI for out-of-dialog requests, they'll push a 
>    route header and leave the r-uri alone. This allows the original R-URI 
>    and any parameters it contains to get delivered to the UAS.
> 
> Unfortunately, I have to say that I dislike this mechanism intensely,
> as it changes the whole philosophy of how SIP messages are routed.

Well, right now, it happens two ways. So I view it as trying to make it 
consistent.

> 
> On a more practical level, it only deals with one level of address
> mapping, and it's not usable by a UA that doesn't implement the change
> -- whereas I would expect that the optimal mechanism is one that
> tweaks the behavior of the proxy doing the mapping, and requires no
> change from any other system.

If the purpose of this is to communicate the targeted URI from the 
terminating proxy to the UA, I don't see how any mechanism can work 
unless it is supported by that proxy and that UA. Any of the mechanisms 
mentioned or discussed (P-Called-Party-ID, To, History-Info) have this 
property.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Certificate authentication in SIP
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On Jul 3, 2007, at 9:38 PM, Jonathan Rosenberg wrote:

> inline.
>
> Dean Willis wrote:
>
>>> So what? I think thats what ought to happen. I'd like to see some  
>>> specific use cases where this can't work with the edge proxy  
>>> performing the authentication. Keep in mind, we are talking about  
>>> *certificate* authentication; that doesn't (by definition)  
>>> required any kind of pre-arranged secret - only a common root CA.
>> My edge proxy might be provided by MCI, but I might be using a  
>> chat service from another provider overseas. This other provider  
>> might be willing to trust the certificate for authentication  
>> (given that I have a signed cert from MCI, who in turn has a  
>> signed CA cert from a recognized op-level CA). However, the chat  
>> provider might not be willing to just accept P-Asserted-Identity  
>> from MCI as an authentication mechanism, since said provider does  
>> not have a trusted peering arrangement with MCI and therefore  
>> ANYBODY could spoof my P-Asserted-Identity.
>
> In a case where your chat provider has no relationship whatsoever  
> with this edge proxy, for what purpose would you be connecting to  
> it in the first place? You should connect directly to your chat  
> provider offering the service. Then, if you have a cert - great -  
> authenticate to it using mutual TLS.
>

Well, I might be doing "outbound" with my edge proxy in order to get  
some help from it for relaying messaging back through my NAT. Or it  
might be a QoS-controlling proxy ala PacketCable. Or a firewall- 
control-proxy . . .

The chat provider might not just be on the other side of my edge  
proxy, but on the other side of my home serving proxy too. And the  
certificate auth stuff would appear to be able to traverse that proxy  
as well.

--
Dea




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Date: Wed, 04 Jul 2007 08:37:32 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] SIP Identity using Media Path
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Dan,

The mechanism described for ICE seems to provide an alternative to ICE's
password-based mechanism for correlating binding requests with
offer-answer exchanges. I have not seen any motivation for this
alternative.

John 

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: 02 July 2007 19:56
> To: 'SIP Working Group'
> Cc: ietf-rtpsec@imc.org
> Subject: [Sip] SIP Identity using Media Path
> 
> As many of you are aware, the signature created by SIP-Identity
> (RFC4474) breaks if an SBC or B2BUA modifies the SDP in the SIP body.
> Typically these modifications include changing the m/c lines (to
> direct the RTP media through the SBC itself, or through a transcoder),
> or performing other adjustments of the SDP to interwork with bugs and
> features of other SIP networks or other endpoints.
> 
> 
> I just submitted a draft, "SIP Identity using Media Path", which
> describes a mechanism that allows RFC4474-like signatures and also
> allows SBCs and B2BUAs to modify the message's SDP.  Links to HTML
> and plain text versions of the Internet Draft are below.
> 
> Abstract:
> 
>    The existing SIP identity mechanism (RFC4474) creates a signature
>    over the SIP body, including the entire SDP.  As part of 
> their normal
>    operation, Session Border Controllers (SBCs) and SIP Back-to-Back
>    User Agents (B2BUAs) modify various fields in the SDP, breaking the
>    signature.
> 
>    This document defines a new mechanism to securely identify the
>    originator of a SIP message while also allowing modification of the
>    SDP by SBCs and B2BUAs.  This new mechanism creates a 
> signature over
>    certain SIP headers and certain SDP lines.  Proof of identity over
>    the media path using DTLS, TLS, HIP, and an extension to ICE are
>    described.
> 
> Please send comments on this draft to the SIP mailing list at
> sip@ietf.org.
> 
> -d
> 
> -----
> 
> HTML version: 
> http://tinyurl.com/25539z
> http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-si
p-identity-med
> ia-00.html
> 
> plain text version: 
> http://tinyurl.com/2gldnv
> http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-si
p-identity-med
> ia-00.txt
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Date: Wed, 04 Jul 2007 16:09:13 +0800
From: Qian Sun <sunqian@huawei.com>
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Hi, 
I am not sure whether the following description in this draft is enough for the usage of Entity-tags with filtering.

      "5.1.  Generating Entity-tags
      The views might correspond to different groups of users that have
      varying levels of access rights to the resource state, or to
      subscribers that have modified their subscription using event
      notification filtering [11]."

My consideration is:
If a subscriber has a subscription with filter F1, the current corresponding entity-tag is E1. Then the same subscriber MUST NOT send SUBSCRIBE with a new filter F2 and the entity-tag E1 in the conditional header field.
i.e., if a subscriber changes filter for an event state, the Suppress-XXX-If-Match should not be used.
The client of subscriber has to record the last filter for event state and compare filters.

Cheers,
Qian




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From: Qian Sun <sunqian@huawei.com>
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Hi,
 
  Figure 2: Example flow (or)*of* a REFER request containin*g* multiple REFER-                               Targets

Cheers,
Qian






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From: "Dan Wing" <dwing@cisco.com>
To: "'Elwell, John'" <john.elwell@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] SIP Identity using Media Path
Date: Wed, 4 Jul 2007 01:45:34 -0700
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> The mechanism described for ICE seems to provide an 
> alternative to ICE's
> password-based mechanism for correlating binding requests with
> offer-answer exchanges.

It's not an alternative to ICE's username and password; the new 
attributes (PK-CHALLENGE, PK-RESPONSE) would be used in addition 
to the existing STUN attributes used by ICE's connectivity
checks.  The ICE username/password would still be used for
message integrity and to associate a flow with an endpoint's
SDP answer.

> I have not seen any motivation for this
> alternative.

The motivation is proving you know the private key 
associated with the public key.  It's only after that
proof that you can be assured the Invite wasn't captured
by an attacker, had its SDP replaced with the attacker's
IP address, and replayed to you.

All of the proof of identity techniques described in the
paper exchange a public key over the media path (HIP, DTLS-SRTP,
TLS) or in the SIP signaling path (ICE).  Then, over the media 
path, the remote party proves he knows the private key 
associated with that public key.  This proof of knowledge of
the private key is part of the handshake for HIP, DTLS,
and TLS; what the paper does is describe how ICE can do 
something similar.  The primary purpose of ICE in that
family of proof of identity techniques is to provide a 
modicum of SBC-survivable identity if RTP (or SRTP 
with Security Descriptions) is used.  Things are obviously
much more secure if DTLS-SRTP, TLS, or HIP are used,
as this ensures nothing is going to inject RTP packets
into the stream; using RTP can't ensure that.


(Crypto experts will point out there is a mafia fraud 
attack problem with the ICE technique as described 
in the paper.  That is true, but if you're not going 
to run SRTP using a secure keying mechanism you're 
suffering the same risk of someone injecting RTP into
your stream as the mafia fraud attack.)

-d


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Subject: Re: [Sip] SIP Identity using Media Path
From: Hans Persson <hasse@ingate.com>
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m=E5n 2007-07-02 klockan 11:55 -0700 skrev Dan Wing:

> I just submitted a draft, "SIP Identity using Media Path", which
> describes a mechanism that allows RFC4474-like signatures and also
> allows SBCs and B2BUAs to modify the message's SDP.  Links to HTML
> and plain text versions of the Internet Draft are below.

It's nice to see work like this that tries to handle the reality of
installed networks instead of pretending that SBCs don't exist.

I'm not sure that I grasp this completely yet, but I'll let it simmer
for a while and give it another read later.


=3D=3D=3D Comments =3D=3D=3D

Section 3, figure 1:

This figure is a simplification, since many enterprises will have their
own SBCs on their borders:

    <--- Enterprise-A ---->
    clients --- PBX --- SBC --- SP1-SBC-1 --- SP1-SBC2 ----+
                                                           |
    +------------------------------------------------------+
    |
    |                              <--- Enterprise-B ---->
    +--- SP2-SBC1 --- SP2-SBC2 --- SBC --- PBX --- clients


Section 3, "mailto URLs" para:

sp1.net and sp2.net (and, later in the document, sp3.net) are (or at
least could be) real domains that shouldn't be used.


Section 4.1, item about Contact:

How can the Contact header usefully be used in the signing process? An
SBC along the message path will happily replace it.



=3D=3D=3D Editorial nits =3D=3D=3D

In the text, RFC references are sometimes written "RFCxxxx" and
sometimes "RFC xxxx". Decide on one alternative.


Section 3, figure 1:

> +----[SP1-SBC1]-[SP1-SBC1]---[SP2-SBC1]-[SP2-SBC2]----+

SP1 has two SBC1. One of them should be SBC2.


Section 3, first para after figure 1:

> using a mailto URL (e.g., 'sip:john.doe@example.com')

Both here and a couple of paragraphs below (under "Mailto URLs:"), sip:
URLs are refered to as mailto URLs which sounds very strange.


Section 3, "mailto URLs" para:

> SP1 would validate the RFC4474 signature modify the SDP.

I can't make sense of this sentence. What should it mean?

> So that a
>      new RFC4474 signature can be created using its own public/private
>      key pair, SP1 needs to modify the From:  field.=20

Rewrite this sentence to

>> SP1 needs to modify the From: field so that a new RFC4474 signature
>> can be created using its own public/private key pair.

> Unlike with E.164
>      numbers which are globally unique, the SP1 isn't able to

Remove the "the" in front of "SP1".


Section 4, second list item:

>    o  This new header is signed in addition to the same set of SIP
>       headers signed by RFC4474 (detailed in )

Detailed in what?


Section 4, para before figure 2:

> In practice, if the

The rest of this sentence has gotten lost somewhere. Please contact the
information desk.


Section 4, step 1:

> Step 1:  Originating endpoint prepares to send an Invite and chooses

"INVITE"

> then sends the Invite to its local SIP proxy.

"INVITE"


Section 4, step 2:

> into the new Identity-Media header.  The invite is forwarded

"INVITE"


Section 4, step 4:

> the Invite.  It validates the Identity-Media signature is
>            valid and was validates it was generated by the originating

"INVITE"

Insert "that" after "It validates". Change "was validates" to "validates
that".


Section 4, step 5:

> authentication service forwards the Invite to the endpoint.

"INVITE"


Section 4, step 6:

> challenge technique from the Invite, and performs that

"INVITE"


Section 4, step 7:

> public key.  The terminating endpoint validates the

Insert "that" after "validates".


Section 4, step 8:

> Step 8:  You can reliably and honestly indicate calling party
>            information ("caller-id") to the terminating endpoint, and
>            ring their phone.

Rewrite to=20

>> Step 8:  The terminating endpoint can reliably and honestly indicate
>>          calling party information ("caller-id") and ring the phone.


Section 4.1, first para:

> "fingerprint").  The specific SDP attribute that are signed depends

"attributes"


Section 4.1, second para:

> The SIP headers that are signed are signed the same as done by

Remove the second "signed".

> the body is not signed.  They signed headers are:

"They" -> "The"


Section 4.1, item about Date:

> the weekday and month items case set as shown in BNF in RFC 3261.

Insert "the" before "BNF".


Section 4.2:

> The authentication service examines the SIP message body for the
>   application/sdp Content-Type.  For all such content-types found, the

"all such content-types" -> "all such message bodies"


Section 5, para 3:

> or HIP session and validates the certificate fingerprint presented in
> SIP signaling matches the certificate presented in the TLS, DTLS,

Insert "that" after "validates" and insert "the" before "SIP".


Section 5.2:

Newline missing after SDP example.


Section 5.3.1:

> subsequent PK-CHALLANGE and PK-RESPONSE, in its SDP.  The syntax is a

"CHALLENGE"


Section 5.3.2.1:

> 5.3.2.1.  PK-CHALLANGE

"CHALLENGE"


Section 7.1:

Write the list items as complete sentences, i.e. start with capitals and
remove ", or" from the end of the first item.

> the device to manufacture a new self-signed certificate (or public

Insert "needs" after "the device".


Section 10.1:

> This example shows how two a=3Dfingerprint lines in SDP would populate
>   a the Media-Fingerprints SIP header field.  The following is an
>   example of an Invite created by the endpoint.

"a the" -> "the"

"INVITE"


Hans

--=20
Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
Ingate Systems AB  +46 13 210857   http://www.ingate.com/

Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/


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Subject: RE: [Sip] SIP Identity using Media Path
From: Hans Persson <hasse@ingate.com>
To: Dan Wing <dwing@cisco.com>
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m=E5n 2007-07-02 klockan 16:04 -0700 skrev Dan Wing:

> > Chapter 9 says:
> >=20
> > "in order for the mechanism to work, SBC-type-of-entities=20
> > must permit DTLS, TLS, ICE, or HIP messages to be exchanged=20
> > in the media path."
> >=20
> > A small question for clarification: at what point must this=20
> > exchange (two-way, I assume) work? As soon as the UAS has=20
> > received the INVITE?
>=20
> Yes.

I think this will have a much better chance of succeeding if the
specification requires this exchange to use the same source and
destination ports as the media that will later be sent on the same
channel. If not, an SBC may lock onto this stream and reject media from
other sources.

Hans

--=20
Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
Ingate Systems AB  +46 13 210857   http://www.ingate.com/

Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/


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Subject: [Sip] Text for the SIP Location Conveyance Security Consideration
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<font face="Courier New">Hi James, <br>
Hi Brian, <br>
<br>
below you can find a first attempt to update the security consideration
section of <br>
<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-ietf-sip-location-conveyance-07">http://tools.ietf.org/html/draft-ietf-sip-location-conveyance-07</a><br>
<br>
Ciao<br>
Hannes<br>
<br>
---------------<br>
<br>
<br>
Security Considerations<br>
<br>
Conveyance of geographical location raises privacy concerns, <br>
and depending on use, there may be confidentiality and access control
aspects. &nbsp;<br>
<br>
This section lists scenarios with their corresponding security <br>
and privacy considerations. <br>
<br>
<br>
Conveying Location by Value <br>
<br>
<br>
&nbsp;&nbsp;&nbsp; UA Alice&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP Proxy&nbsp;&nbsp; UAS-A&nbsp;&nbsp; UAS-B&nbsp;&nbsp; UAS-C<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; SIP Message&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; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |----------------------------&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; Location (value/body)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp;&nbsp;&nbsp;&nbsp; |-----------------&gt;|<br>
&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;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP Message&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;-----------------------------------------------|<br>
&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Figure z: Location by Value<br>
<br>
Figure z shows a SIP message that carries an LbyV in the body of the SIP<br>
message.<br>
<br>
Conveying a Reference to Location Information <br>
<br>
<br>
&nbsp;&nbsp;&nbsp; UA Alice&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP Proxy&nbsp;&nbsp; UAS-A&nbsp;&nbsp; UAS-B&nbsp;&nbsp; UAS-C<br>
&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; Location Server <br>
&nbsp;&nbsp;&nbsp; |&nbsp; SIP Message&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; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp; |----------------------------&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp; |&nbsp; SIP Message + LbyR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp; |-------------------------&gt;|<br>
&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;&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; |<br>
&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;&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; |<br>
&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;&nbsp; |&nbsp; Resolve LbyR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp; |&lt;=========================|<br>
&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;&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; |<br>
&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;&nbsp; |&nbsp; PIDF-LO&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp; |=========================&gt;|<br>
&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;&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; |<br>
&nbsp;&nbsp;&nbsp; |&nbsp; SIP Message&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp; |&lt;-------------------------------------------------------|<br>
&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;&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; |<br>
&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;&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; |<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Figure x: Usage Scenario: Proxy adding LbyR<br>
<br>
<br>
Figure x shows an exchange where a SIP proxy acts as a Location Server
and <br>
attaches a LbyR to the header of the bypassing SIP message. A separate<br>
protocol interaction is needed to resolve the reference to a PIDF-LO. <br>
<br>
<br>
&nbsp;&nbsp; UA Alice&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; LCS&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UAS-A&nbsp;&nbsp; UAS-B&nbsp;&nbsp; UAS-C<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; Request LbyR&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |----------------------------&gt;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; LbyR&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;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&lt;----------------------------|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; SIP Message including LbyR&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; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |-------------------------------------------------------&gt;|<br>
&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;&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; |<br>
&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;&nbsp;&nbsp; | Resolve LbyR&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp;&nbsp; |&lt;=========================|<br>
&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;&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; |<br>
&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;&nbsp;&nbsp; | PIDF-LO&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&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;&nbsp;&nbsp; |=========================&gt;|<br>
&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;&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; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp; SIP Message&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<br>
&nbsp;&nbsp;&nbsp;&nbsp; |&lt;-------------------------------------------------------|<br>
&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;&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; |<br>
&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;&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; |<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Figure y: Usage Scenario: End Host adding Reference<br>
<br>
Figure y depicts a message exchange where an LbyR was obtained outside
the <br>
SIP protocol exchange, in this case using the Location Configuration
Protocol, and <br>
where the LbyR is attached to the header or the body of the SIP
message. <br>
<br>
Security Guidelines<br>
<br>
* Determining the Location Recipient<br>
<br>
An end point should know whether location information (by value or by
reference) will be used for location-based routing or for consumption
by the other communication end point. This applies to Figure x, y and
z. When location information is meant for consumption by the other
communication end point only then the usage of S/MIME is RECOMMENDED to
protect the LbyV and LbyR against eavesdropping. The values MUST be
placed into the body of the message. Alternatively, in some
environments hop-by-hop security using TLS MAY provide sufficient
security protection (depending on the location-based application). <br>
Since retargeting and forking might prevent the communication initiator
to obtain the correct public key it is suggested to convey the LbyV /
LbyR after the initial communication setup, for example, using an
UPDATE message. To simplify the management of certificates the usage of
SIP CERT is suggested, see [I-D.ietf-sip-cert]. <br>
<br>
For cases where location-based routing is envisioned S/MIME cannot be
used since the entity performing the location based routing interaction
is likely not to be known in advance. Hence, TLS MUST be used to
protect the LO in transit.<br>
<br>
In cases where a proxy adds an LbyR TLS MUST be used from the SIP proxy
towards the final recipient. <br>
<br>
* Computing a LbyR<br>
<br>
To prevent others from accessing the LbyR for a particular entity, the
reference to a Location Object MUST NOT be<br>
guessable. It is RECOMMENDED that the reference contains a random
component with at least 128 bit of entropy. For providing randomness
the best current practice guidelines in RFC 4086 [RFC4086] have to be
applied. It is furthermore mandated to use a SIPS URI schema, which
implies the usage of TLS with confidentiality protection, to prevent
eavesdroppers to observe the protocol exchange between the Location
Recipient and the Location Server when dereferencing the LbyR. <br>
<br>
These recommendations refer only to Figure x where the SIP proxy adds
the reference to the outgoing message. With the scenario in Figure y
the acquisition of the LbyR and subsequent conveyance is decoupled and
hence it is the responsibility of the LCP specification and ultimately
of the LCS to provide a reference with the same randomness. <br>
<br>
It is not necessary to digitally sign the PIDF-LO since authentication
of Location Server is provided by the dereferencing protocol. <br>
<br>
* Content of the PIDF-LO<br>
<br>
The PIDF-LO contains identity information and privacy rules. In cases
where the end host adds a the PIDF-LO the content of the identity and
the rules conveyed to the Location Recipient are under full control by
the end point itself. However, when a SIP proxy adds a reference to
location information it needs to guess the context. There are, however,
a few information elements in the SIP message itself that provide hint
about the desirable behavior by the SIP proxy.<br>
<br>
When SIP privacy mechanisms, see RFC 3323, are used then the PIDF-LO
MUST NOT contain the identity of Target in the entity field of the PIDF
document. Furthermore, the extended privacy rules and the Note Well
statement MUST NOT provide linkability with the identity of the Target.<br>
<br>
When the SIP proxy adds a LbyR and does not know the privacy policies
of the Target then it MUST set to the values to the following values: <br>
<br>
* The note-well element MUST NOT contain human readable privacy
policies. <br>
* The retransmission-allowed element is set to zero (0), i.e., further
distribution<br>
&nbsp;&nbsp; is not allowed. <br>
* The retention-expires field MUST be set to a maximum of 24 hours.<br>
* The extended policy rules MUST not be present.<br>
* The entity field in the PIDF document MUST be empty.<br>
<br>
This document does not provide information how the current context of
the communication initiator is determined other than by inspecting
information provided by RFC 3323 and implicitly via the Service URN
[I-D.ietf-ecrit-service-urn]. At the time of writing this specification
this functionality is made available within the SIP SAML
[I-D.ietf-sip-saml] specification. <br>
<br>
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] SIP Identity using Media Path
In-reply-to: <1d5f01c7be17$b07aedc0$2014590a@amer.cisco.com>
To: Dan Wing <dwing@cisco.com>, sip@ietf.org
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Dan,

Thanks for the explanation. However, I am still not certain I have quite
"got it" yet. Let me explain what I think it is doing, and you can
confirm or deny.

A sends an INVITE request with SDP offer containing ICE candidates and a
password. A's local proxy adds the Identity-Media header field. B
receives the INVITE request and knows, from the From and Identity-Media
header fields, that it has comes from A, and therefore that the password
has come from A. So if nobody has been able to eavesdrop on the INVITE
request, this gives proof of ownership of STUN requests demonstrating
possession of that password. Right?

However, the assumption that there is no eavesdropping is not a safe
assumption (even if SIPS is use, I think we are all familiar with its
limitations). By conveying a public key rather than a password,
eavesdropping is not a problem. Right?

More below.

John

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: 04 July 2007 09:46
> To: Elwell, John; sip@ietf.org
> Cc: ietf-rtpsec@imc.org
> Subject: RE: [Sip] SIP Identity using Media Path
> 
> > The mechanism described for ICE seems to provide an 
> > alternative to ICE's
> > password-based mechanism for correlating binding requests with
> > offer-answer exchanges.
> 
> It's not an alternative to ICE's username and password; the new 
> attributes (PK-CHALLENGE, PK-RESPONSE) would be used in addition 
> to the existing STUN attributes used by ICE's connectivity
> checks.  The ICE username/password would still be used for
> message integrity and to associate a flow with an endpoint's
> SDP answer.
> 
> > I have not seen any motivation for this
> > alternative.
> 
> The motivation is proving you know the private key 
> associated with the public key.  It's only after that
> proof that you can be assured the Invite wasn't captured
> by an attacker, had its SDP replaced with the attacker's
> IP address, and replayed to you.
[JRE] The Identity header field would prevent this (although the
Identity-Media header field does not).

> 
> All of the proof of identity techniques described in the
> paper exchange a public key over the media path (HIP, DTLS-SRTP,
> TLS) or in the SIP signaling path (ICE).  Then, over the media 
> path, the remote party proves he knows the private key 
> associated with that public key.  This proof of knowledge of
> the private key is part of the handshake for HIP, DTLS,
> and TLS; what the paper does is describe how ICE can do 
> something similar.  The primary purpose of ICE in that
> family of proof of identity techniques is to provide a 
> modicum of SBC-survivable identity if RTP (or SRTP 
> with Security Descriptions) is used.  Things are obviously
> much more secure if DTLS-SRTP, TLS, or HIP are used,
> as this ensures nothing is going to inject RTP packets
> into the stream; using RTP can't ensure that.
> 
> 
> (Crypto experts will point out there is a mafia fraud 
> attack problem with the ICE technique as described 
> in the paper.  That is true, but if you're not going 
> to run SRTP using a secure keying mechanism you're 
> suffering the same risk of someone injecting RTP into
> your stream as the mafia fraud attack.)
> 
> -d
> 


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Subject: RE: [Sip] Security issues with ietf-sip-loop-fix
Date: Wed, 4 Jul 2007 13:07:48 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Cullen Jennings" <fluffy@cisco.com>, "SIP" <sip@ietf.org>
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(as individual)

With regard to:

> If we go forward with path 1, the value of this draft is=20
> probably not super high because Max-Forwards deals with the=20
> the accidental loops in most cases  (but not all cases). If=20
> we go with path 2, the WG will need make sure it OK with the=20
> solution. I'm certainly open to other paths forward if=20
> someone has a good idea.

There have been comments previously on this list that the default value
of 70 for Max-Forwards is too high, and that a lower default value
should have been used. Therefore Max-Forwards will find the accidental
loops, but will take longer to do it, if the default values are used.

Regards

Keith=20

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: Tuesday, July 03, 2007 3:36 AM
> To: SIP
> Subject: [Sip] Security issues with ietf-sip-loop-fix
>=20
>=20
> (this email with my AD hat on)
>=20
> The type of SIP loops described in ietf-sip-loop-fix can be=20
> roughly classified into two categories. Loops that had=20
> happened because an accidental registration that causes a=20
> loop, and loops that happened due to malicious intent.
>=20
> This draft attempted to deals with both. The abstract describes it as
>=20
>    This document normatively updates RFC 3261, the Session Initiation
>     Protocol (SIP), to address a security vulnerability=20
> identified in SIP
>     proxy behavior.  This vulnerability enables an attack against SIP
>     networks where a small number of legitimate, even authorized, SIP
>     requests can stimulate massive amounts of proxy-to-proxy traffic.
>=20
> During IETF LC, substantial comment came up about the=20
> security of this draft not really working in the case of=20
> malicious loops. A thread ensues that convinced many people=20
> on the thread that this draft did not really help with the=20
> malicious loops. (Not 100% sure here but I suspect that the=20
> authors and chairs were all convinced of this). I will let=20
> others explain the issues that came up as we decide what to do next.
>=20
> I think that some of issue was discussed while the working=20
> group was considering this draft - however, I suspect that=20
> many participants did not understand how easily this security=20
> would be defeated. I for one significantly underestimated the=20
> how easy it was for a malicious attacker to cause this=20
> problem. I apologize to the folks that attempted to explain=20
> it to me and I just failed to "get it".
>=20
> Anyways, we need to figure out what to do next with this=20
> draft. There are basically two paths forward that I see:
>=20
> 1) reduce the scope so that it only deals with accidental=20
> loops and provides no protection against malicious loops. I=20
> don't think this would change any of the mechanisms in the=20
> draft but it would probably change lots of the text around=20
> what security claims the draft made.
>=20
> 2) try to extend or replace the mechanism with one that does=20
> protect against malicious attack. A strong candidate to=20
> consider would be the work in draft-sparks-sipping-max-breadth.
>=20
> If we go forward with path 1, the value of this draft is=20
> probably not super high because Max-Forwards deals with the=20
> the accidental loops in most cases  (but not all cases). If=20
> we go with path 2, the WG will need make sure it OK with the=20
> solution. I'm certainly open to other paths forward if=20
> someone has a good idea.
>=20
> What I would like to do now is start some discussion about=20
> what the working group would like to do with this draft.
>=20
> Thanks, Cullen
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jul 04 07:11:50 2007
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Subject: RE: [Sip] Security issues with ietf-sip-loop-fix
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Cullen Jennings" <fluffy@cisco.com>, "SIP" <sip@ietf.org>
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(As WG chair and document shepherd for this document)

We do need to get this resolved, and have therefore allocated a slot to
the discussion at the forthcoming IETF meeting of the SIP WG.

This does however require material to put on the slides to lead the
discussion, rather than spending 15 minutes drawing thoughts out of the
microphone stream at the meeting.

So please post to the list with views on this before the meeting,
ideally backed up with technical thoughts to justify your position.

Regards

Keith=20

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: Tuesday, July 03, 2007 3:36 AM
> To: SIP
> Subject: [Sip] Security issues with ietf-sip-loop-fix
>=20
>=20
> (this email with my AD hat on)
>=20
> The type of SIP loops described in ietf-sip-loop-fix can be=20
> roughly classified into two categories. Loops that had=20
> happened because an accidental registration that causes a=20
> loop, and loops that happened due to malicious intent.
>=20
> This draft attempted to deals with both. The abstract describes it as
>=20
>    This document normatively updates RFC 3261, the Session Initiation
>     Protocol (SIP), to address a security vulnerability=20
> identified in SIP
>     proxy behavior.  This vulnerability enables an attack against SIP
>     networks where a small number of legitimate, even authorized, SIP
>     requests can stimulate massive amounts of proxy-to-proxy traffic.
>=20
> During IETF LC, substantial comment came up about the=20
> security of this draft not really working in the case of=20
> malicious loops. A thread ensues that convinced many people=20
> on the thread that this draft did not really help with the=20
> malicious loops. (Not 100% sure here but I suspect that the=20
> authors and chairs were all convinced of this). I will let=20
> others explain the issues that came up as we decide what to do next.
>=20
> I think that some of issue was discussed while the working=20
> group was considering this draft - however, I suspect that=20
> many participants did not understand how easily this security=20
> would be defeated. I for one significantly underestimated the=20
> how easy it was for a malicious attacker to cause this=20
> problem. I apologize to the folks that attempted to explain=20
> it to me and I just failed to "get it".
>=20
> Anyways, we need to figure out what to do next with this=20
> draft. There are basically two paths forward that I see:
>=20
> 1) reduce the scope so that it only deals with accidental=20
> loops and provides no protection against malicious loops. I=20
> don't think this would change any of the mechanisms in the=20
> draft but it would probably change lots of the text around=20
> what security claims the draft made.
>=20
> 2) try to extend or replace the mechanism with one that does=20
> protect against malicious attack. A strong candidate to=20
> consider would be the work in draft-sparks-sipping-max-breadth.
>=20
> If we go forward with path 1, the value of this draft is=20
> probably not super high because Max-Forwards deals with the=20
> the accidental loops in most cases  (but not all cases). If=20
> we go with path 2, the WG will need make sure it OK with the=20
> solution. I'm certainly open to other paths forward if=20
> someone has a good idea.
>=20
> What I would like to do now is start some discussion about=20
> what the working group would like to do with this draft.
>=20
> Thanks, Cullen
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Elwell, John" <john.elwell@siemens.com>
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Eric,

I would just like to correct a slight inaccuracy in this draft. RFC 3372
does not specify the use of INFO for QSIG, but RFC 3204 (which is
referenced by RFC 3372) does define the MIME type for QSIG. The
transport of QSIG using the SIP INFO method is instead specified in
Standard ECMA-355 from Ecma-International. This is indeed a very similar
usage of INFO to that specified in RFC 3372 and should be allowed in the
proposed update to RFC 2976.

John 

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org] 
> Sent: 03 July 2007 19:15
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-burger-sip-info-00.txt
> 
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> 
> 
> 	Title		: Session Initiation Protocol (SIP) 
> INFO Method Context
> 	Author(s)	: E. Burger
> 	Filename	: draft-burger-sip-info-00.txt
> 	Pages		: 8
> 	Date		: 2007-7-3
> 	
>    The purpose of the INFO request for the Session Initiation Protocol
>    (SIP), as described by RFC 2976, is to provide mid-session SIP User
>    Agent (UA)-to-SIP UA application data transport.  In the 
> years since
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From: "GOLDMAN, STUART O \(STUART\)" <sgoldman@alcatel-lucent.com>
To: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
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Folks,

=20

It does seem useful to be able to include the RPH in the response
messages, and I don't see any downside in allowing this.

=20

Stuart Goldman

Alcatel-Lucent=20

sgoldman@alcatel-lucent.com

602 493 8438

=20

=20

________________________________

From: DOLLY, MARTIN C, ATTLABS [mailto:mdolly@att.com]=20
Sent: Friday, June 29, 2007 8:29 AM
To: Janet P Gunn; James M. Polk
Cc: sip@ietf.org
Subject: RE: [Sip] New draft modifying RFC 4412 for Responses

=20

Hello,

=20

Agreed. In addition, not having the RPH in the responses would mean the
proxies would have to maintain the state of the RPH value.

=20

Martin

=20

________________________________

From: Janet P Gunn [mailto:jgunn6@csc.com]=20

In working on specifying and deploying systems that will actually use
the RPH with the ets and wps namespaces, we have found several contexts
in which the RPH  on the response is very useful.=20

1)        Error reporting.  RFC 4412 does not give clear direction on
the treatment of RPH with syntax errors (out of range priority values,
duplicate namespaces, etc.)  We have found that including the RPH in the
4xx responses is usesful in addressing the errors.=20
2)        In our system, we often set the ets  RPH (with default
priority)  based on the "ETS-DN""dialed number" or uri, before
authentication and before knowing the correct priority value.  Including
the RPH in the response is a useful way of conveying the correct
priority value back to nodes earlier in the "call flow".=20
3)        Similarly, if an RP actor receives a SIP INVITE with the
ETS-DN, but no RPH it may, after authenticating the call, include RPH in
the responses in order to inform  nodes earlier in the call flow that
this is a priority call.=20
4)        If the RPH is being used to prioritize processing (of both
messages and responses) in the SIP user agent, including the RPH in the
response allows the receiving RP actor to give the response "ETS
treatment" before checking on the "state" of the call.  This is
particularly relevant when the RP actor itself is congested.=20

Janet


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<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Comic Sans =
MS"><span
style=3D'font-size:12.0pt;font-family:"Comic Sans =
MS";color:blue'>Folks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Comic Sans =
MS"><span
style=3D'font-size:12.0pt;font-family:"Comic Sans =
MS";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Comic Sans =
MS"><span
style=3D'font-size:12.0pt;font-family:"Comic Sans MS";color:blue'>It =
does seem
useful to be able to include the RPH in the response messages, and I =
don&#8217;t
see any downside in allowing this.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Comic Sans =
MS"><span
style=3D'font-size:12.0pt;font-family:"Comic Sans =
MS";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoAutoSig><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>Stuart =
Goldman<o:p></o:p></span></font></p>

<p class=3DMsoAutoSig><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>Alcatel-Lucent =
<o:p></o:p></span></font></p>

<p class=3DMsoAutoSig><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>sgoldman@alcatel-lucent.com<o:p></o=
:p></span></font></p>

<p class=3DMsoAutoSig><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>602 493 =
8438<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;color:blue'><o:p>&nbsp;</o:p></span></font></p>=


</div>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Comic Sans =
MS"><span
style=3D'font-size:12.0pt;font-family:"Comic Sans =
MS";color:blue'><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'> =
DOLLY, MARTIN C,
ATTLABS [mailto:mdolly@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Friday, June 29, =
2007 8:29
AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Janet P Gunn; James =
M. Polk<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] New =
draft
modifying RFC 4412 for Responses</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=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Hello,</span></font><o:p></o:p></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'>Agreed. In addition, not having the =
RPH in
the responses would mean the proxies would have to maintain the state of =
the
RPH value.</span></font><o:p></o:p></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'>Martin</span></font><o:p></o:p></p>

<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>

<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'> Janet =
P Gunn
[mailto:jgunn6@csc.com] </span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>In working on =
specifying
and deploying systems that will actually use the RPH with the ets and =
wps
namespaces, we have found several contexts in which the RPH &nbsp;on the
response is very useful. <br>
<br>
1) &nbsp; &nbsp; &nbsp; &nbsp;Error reporting. &nbsp;RFC 4412 does not =
give
clear direction on the treatment of RPH with syntax errors (out of range
priority values, duplicate namespaces, etc.) &nbsp;We have found that =
including
the RPH in the &nbsp;4xx responses is usesful in addressing the errors. =
<br>
2) &nbsp; &nbsp; &nbsp; &nbsp;In our system, we often set the ets =
&nbsp;RPH (with
default priority) &nbsp;based on the &#8220;ETS-DN&#8221;&#8220;dialed
number&#8221; or uri, before authentication and before knowing the =
correct
priority value. &nbsp;Including the RPH in the response is a useful way =
of
conveying the correct priority value back to nodes earlier in the =
&#8220;call
flow&#8221;. <br>
3) &nbsp; &nbsp; &nbsp; &nbsp;Similarly, if an RP actor receives a SIP =
INVITE
with the ETS-DN, but no RPH it may, after authenticating the call, =
include RPH
in the responses in order to inform &nbsp;nodes earlier in the call flow =
that
this is a priority call. <br>
4) &nbsp; &nbsp; &nbsp; &nbsp;If the RPH is being used to prioritize =
processing
(of both messages and responses) in the SIP user agent, including the =
RPH in
the response allows the receiving RP actor to give the response =
&#8220;ETS
treatment&#8221; before checking on the &#8220;state&#8221; of the call.
&nbsp;This is particularly relevant when the RP actor itself is =
congested. <br>
<br>
</span></font><font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;
font-family:sans-serif'>Janet</span></font><o:p></o:p></p>

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> > I just submitted a draft, "SIP Identity using Media Path", which
> > describes a mechanism that allows RFC4474-like signatures and also
> > allows SBCs and B2BUAs to modify the message's SDP.  Links to HTML
> > and plain text versions of the Internet Draft are below.
> 
> It's nice to see work like this that tries to handle the reality of
> installed networks instead of pretending that SBCs don't exist.

Thanks.

> I'm not sure that I grasp this completely yet, but I'll let it simmer
> for a while and give it another read later.
> 
> 
> === Comments ===
> 
> Section 3, figure 1:
> 
> This figure is a simplification, since many enterprises will 
> have their
> own SBCs on their borders:
> 
>     <--- Enterprise-A ---->
>     clients --- PBX --- SBC --- SP1-SBC-1 --- SP1-SBC2 ----+
>                                                            |
>     +------------------------------------------------------+
>     |
>     |                              <--- Enterprise-B ---->
>     +--- SP2-SBC1 --- SP2-SBC2 --- SBC --- PBX --- clients


True.  I'll figure out a clearer way to diagram that.

> Section 3, "mailto URLs" para:
> 
> sp1.net and sp2.net (and, later in the document, sp3.net) are (or at
> least could be) real domains that shouldn't be used.

Agreed; I may go with "sp1.example.net" and "sp2.example.net", 
although this is less clear than using unique top-level domain 
names.

> Section 4.1, item about Contact:
> 
> How can the Contact header usefully be used in the signing process? An
> SBC along the message path will happily replace it.

I have removed that in -01 (which isn't yet published, of course).  There
was some thought it was necessary, but I agree it should be removed
from the signature.

On a similar note, I am considering removing CallId from the signature.
Oftentimes the Call Id value contains an IP address (in dotted decimal
or hex), and an SBC or B2BUA may also want to rewrite such a CallId.  I
have made a note of that in -01 so this can be discussed.

> === Editorial nits ===

Thanks for the list of nits - I'll get those cleaned up in -01.

-d


> In the text, RFC references are sometimes written "RFCxxxx" and
> sometimes "RFC xxxx". Decide on one alternative.
> 
> 
> Section 3, figure 1:
> 
> > +----[SP1-SBC1]-[SP1-SBC1]---[SP2-SBC1]-[SP2-SBC2]----+
> 
> SP1 has two SBC1. One of them should be SBC2.
>
> Section 3, first para after figure 1:
> 
> > using a mailto URL (e.g., 'sip:john.doe@example.com')
> 
> Both here and a couple of paragraphs below (under "Mailto 
> URLs:"), sip:
> URLs are refered to as mailto URLs which sounds very strange.
>
> Section 3, "mailto URLs" para:
> 
> > SP1 would validate the RFC4474 signature modify the SDP.
> 
> I can't make sense of this sentence. What should it mean?
>
> > So that a
> >      new RFC4474 signature can be created using its own 
> public/private
> >      key pair, SP1 needs to modify the From:  field. 
> 
> Rewrite this sentence to
> 
> >> SP1 needs to modify the From: field so that a new RFC4474 signature
> >> can be created using its own public/private key pair.
>
> > Unlike with E.164
> >      numbers which are globally unique, the SP1 isn't able to
> 
> Remove the "the" in front of "SP1".
> 
> 
> Section 4, second list item:
> 
> >    o  This new header is signed in addition to the same set of SIP
> >       headers signed by RFC4474 (detailed in )
> 
> Detailed in what?
> 
> 
> Section 4, para before figure 2:
> 
> > In practice, if the
> 
> The rest of this sentence has gotten lost somewhere. Please 
> contact the
> information desk.
> 
> 
> Section 4, step 1:
> 
> > Step 1:  Originating endpoint prepares to send an Invite and chooses
> 
> "INVITE"
> 
> > then sends the Invite to its local SIP proxy.
> 
> "INVITE"
> 
> 
> Section 4, step 2:
> 
> > into the new Identity-Media header.  The invite is forwarded
> 
> "INVITE"
> 
> 
> Section 4, step 4:
> 
> > the Invite.  It validates the Identity-Media signature is
> >            valid and was validates it was generated by the 
> originating
> 
> "INVITE"
> 
> Insert "that" after "It validates". Change "was validates" to 
> "validates
> that".
> 
> 
> Section 4, step 5:
> 
> > authentication service forwards the Invite to the endpoint.
> 
> "INVITE"
> 
> 
> Section 4, step 6:
> 
> > challenge technique from the Invite, and performs that
> 
> "INVITE"
> 
> 
> Section 4, step 7:
> 
> > public key.  The terminating endpoint validates the
> 
> Insert "that" after "validates".
> 
> 
> Section 4, step 8:
> 
> > Step 8:  You can reliably and honestly indicate calling party
> >            information ("caller-id") to the terminating 
> endpoint, and
> >            ring their phone.
> 
> Rewrite to 
> 
> >> Step 8:  The terminating endpoint can reliably and 
> honestly indicate
> >>          calling party information ("caller-id") and ring 
> the phone.
> 
> 
> Section 4.1, first para:
> 
> > "fingerprint").  The specific SDP attribute that are signed depends
> 
> "attributes"
> 
> 
> Section 4.1, second para:
> 
> > The SIP headers that are signed are signed the same as done by
> 
> Remove the second "signed".
> 
> > the body is not signed.  They signed headers are:
> 
> "They" -> "The"
> 
> 
> Section 4.1, item about Date:
> 
> > the weekday and month items case set as shown in BNF in RFC 3261.
> 
> Insert "the" before "BNF".
> 
> 
> Section 4.2:
> 
> > The authentication service examines the SIP message body for the
> >   application/sdp Content-Type.  For all such content-types 
> found, the
> 
> "all such content-types" -> "all such message bodies"
> 
> 
> Section 5, para 3:
> 
> > or HIP session and validates the certificate fingerprint 
> presented in
> > SIP signaling matches the certificate presented in the TLS, DTLS,
> 
> Insert "that" after "validates" and insert "the" before "SIP".
> 
> 
> Section 5.2:
> 
> Newline missing after SDP example.
> 
> 
> Section 5.3.1:
> 
> > subsequent PK-CHALLANGE and PK-RESPONSE, in its SDP.  The 
> syntax is a
> 
> "CHALLENGE"
> 
> 
> Section 5.3.2.1:
> 
> > 5.3.2.1.  PK-CHALLANGE
> 
> "CHALLENGE"
> 
> 
> Section 7.1:
> 
> Write the list items as complete sentences, i.e. start with 
> capitals and
> remove ", or" from the end of the first item.
> 
> > the device to manufacture a new self-signed certificate (or public
> 
> Insert "needs" after "the device".
> 
> 
> Section 10.1:
> 
> > This example shows how two a=fingerprint lines in SDP would populate
> >   a the Media-Fingerprints SIP header field.  The following is an
> >   example of an Invite created by the endpoint.
> 
> "a the" -> "the"
> 
> "INVITE"
> 
> 
> Hans
> 
> -- 
> Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
> Ingate Systems AB  +46 13 210857   http://www.ingate.com/
> 
> Private: <unicorn@lysator.liu.se>  
> http://www.lysator.liu.se/~unicorn/


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Subject: RE: [Sip] SIP Identity using Media Path
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> m=E5n 2007-07-02 klockan 16:04 -0700 skrev Dan Wing:
>=20
> > > Chapter 9 says:
> > >=20
> > > "in order for the mechanism to work, SBC-type-of-entities=20
> > > must permit DTLS, TLS, ICE, or HIP messages to be exchanged=20
> > > in the media path."
> > >=20
> > > A small question for clarification: at what point must this=20
> > > exchange (two-way, I assume) work? As soon as the UAS has=20
> > > received the INVITE?
> >=20
> > Yes.
>=20
> I think this will have a much better chance of succeeding if the
> specification requires this exchange to use the same source and
> destination ports as the media that will later be sent on the same
> channel.

Yes, that is how HIP's key exchange, TLS, DTLS-SRTP, and ICE
all function.

> If not, an SBC may lock onto this stream and reject=20
> media from other sources.

Yes, I agree that will give the best chance.  Of course, it is
always possible for an SBC (or firewall) to block such messages=20
anyway (as has been discussed on the RTPSEC mailer).

-d


> Hans
>=20
> --=20
> Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
> Ingate Systems AB  +46 13 210857   http://www.ingate.com/
>=20
> Private: <unicorn@lysator.liu.se> =20
> http://www.lysator.liu.se/~unicorn/


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Subject: RE: [Sip] SIP Identity using Media Path
From: Hans Persson <hasse@ingate.com>
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ons 2007-07-04 klockan 09:40 -0700 skrev Dan Wing:

> >     <--- Enterprise-A ---->
> >     clients --- PBX --- SBC --- SP1-SBC-1 --- SP1-SBC2 ----+
> >                                                            |
> >     +------------------------------------------------------+
> >     |
> >     |                              <--- Enterprise-B ---->
> >     +--- SP2-SBC1 --- SP2-SBC2 --- SBC --- PBX --- clients
> 
> True.  I'll figure out a clearer way to diagram that.

Possibly it gets better if done up-down instead of left-right.

> > Section 3, "mailto URLs" para:
> > 
> > sp1.net and sp2.net (and, later in the document, sp3.net) are (or at
> > least could be) real domains that shouldn't be used.
> 
> Agreed; I may go with "sp1.example.net" and "sp2.example.net", 
> although this is less clear than using unique top-level domain 
> names.

Perhaps it would be clearer to use for instance sp1.example.com,
sp2.example.net and sp3.example.org.

> > Section 4.1, item about Contact:
> > 
> > How can the Contact header usefully be used in the signing process? An
> > SBC along the message path will happily replace it.
> 
> I have removed that in -01 (which isn't yet published, of course).  There
> was some thought it was necessary, but I agree it should be removed
> from the signature.
> 
> On a similar note, I am considering removing CallId from the signature.
> Oftentimes the Call Id value contains an IP address (in dotted decimal
> or hex), and an SBC or B2BUA may also want to rewrite such a CallId.  I
> have made a note of that in -01 so this can be discussed.

I thought about that as well, but apparently I forgot about adding it to
my mail. I think it would be a good idea to take Call-ID out as well.

Hans

-- 
Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
Ingate Systems AB  +46 13 210877   http://www.ingate.com/

Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/




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Subject: RE: [Sip] SIP Identity using Media Path
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ons 2007-07-04 klockan 10:01 -0700 skrev Dan Wing:

> Yes, I agree that will give the best chance.  Of course, it is
> always possible for an SBC (or firewall) to block such messages 
> anyway (as has been discussed on the RTPSEC mailer).

But that would at least mean that the SBC would have to look inside the
media stream to see that not everything in it is (S)RTP, instead of just
looking at the source and destination. 

It may happen, of course, but at least it's less likely. When it does
appear, it would require an SBC that can recognize the security
interchange for what it is and allow it to pass even though it isn't
media. I'm not sure if that can reasonably be done.

Hans

-- 
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Ingate Systems AB  +46 13 210877   http://www.ingate.com/

Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/




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Subject: RE: [Sip] SIP Identity using Media Path
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> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Wednesday, July 04, 2007 3:19 AM
> To: Dan Wing; sip@ietf.org
> Cc: ietf-rtpsec@imc.org
> Subject: RE: [Sip] SIP Identity using Media Path
> 
> Dan,
> 
> Thanks for the explanation. However, I am still not certain I 
> have quite
> "got it" yet. Let me explain what I think it is doing, and you can
> confirm or deny.
> 
> A sends an INVITE request with SDP offer containing ICE 
> candidates and a
> password. A's local proxy adds the Identity-Media header field. B
> receives the INVITE request and knows, from the From and 
> Identity-Media
> header fields, that it has comes from A, and therefore that 
> the password
> has come from A. So if nobody has been able to eavesdrop on the INVITE
> request, this gives proof of ownership of STUN requests demonstrating
> possession of that password. Right?

Yes -- you could accomplish exactly that by only validating the
signature over the headers and not doing the HIP/ICE/DTLS-SRTP/TLS
media path validation.

However, this means you have transitive trust of all the SIP proxies
along the path.  It also means you don't think an attacker has captured
a valid, signed Invite, and is replaying it towards you.

> However, the assumption that there is no eavesdropping is not a safe
> assumption (even if SIPS is use, I think we are all familiar with its
> limitations). By conveying a public key rather than a password,
> eavesdropping is not a problem. Right?

Right.  

(Although I have variously convinced myself there is still a mafia
fraud attack problem with the ICE technique, and have convinced myself
there isn't such an attack.  I'll need to leave that analysis until
later when I have sufficient time to write it up.  Today I think there
isn't a problem (because the public key is authenticated by A's 
signature, which prevents a man-in-the-middle from doing a mafia
fraud attack by replacing the A's public key with his own public
key.)

> More below.
> 
> John
> 
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com] 
> > Sent: 04 July 2007 09:46
> > To: Elwell, John; sip@ietf.org
> > Cc: ietf-rtpsec@imc.org
> > Subject: RE: [Sip] SIP Identity using Media Path
> > 
> > > The mechanism described for ICE seems to provide an 
> > > alternative to ICE's
> > > password-based mechanism for correlating binding requests with
> > > offer-answer exchanges.
> > 
> > It's not an alternative to ICE's username and password; the new 
> > attributes (PK-CHALLENGE, PK-RESPONSE) would be used in addition 
> > to the existing STUN attributes used by ICE's connectivity
> > checks.  The ICE username/password would still be used for
> > message integrity and to associate a flow with an endpoint's
> > SDP answer.
> > 
> > > I have not seen any motivation for this
> > > alternative.
> > 
> > The motivation is proving you know the private key 
> > associated with the public key.  It's only after that
> > proof that you can be assured the Invite wasn't captured
> > by an attacker, had its SDP replaced with the attacker's
> > IP address, and replayed to you.
> [JRE] The Identity header field would prevent this (although the
> Identity-Media header field does not).

Right.

To combat this attack, Identity (RFC4474) relies on IP address
and port numbers.  However, NATs, TURN, and SBCs reduce the 
value of the IP address and port numbers as a unique identifier
of a flow before the RFC4474 replay window has expired.

SIP-Identity-Media doesn't use IP addresses -- rather, it 
has the endpoints prove they know the private key associated 
with their public key.  This allows SBCs to rewrite m/c lines,
and also removes the risk, inherient in RFC4474, that an 
attacker will be able to get your recently-used transport 
address assigned to them by the same NAT or TURN server.

-d

> > All of the proof of identity techniques described in the
> > paper exchange a public key over the media path (HIP, DTLS-SRTP,
> > TLS) or in the SIP signaling path (ICE).  Then, over the media 
> > path, the remote party proves he knows the private key 
> > associated with that public key.  This proof of knowledge of
> > the private key is part of the handshake for HIP, DTLS,
> > and TLS; what the paper does is describe how ICE can do 
> > something similar.  The primary purpose of ICE in that
> > family of proof of identity techniques is to provide a 
> > modicum of SBC-survivable identity if RTP (or SRTP 
> > with Security Descriptions) is used.  Things are obviously
> > much more secure if DTLS-SRTP, TLS, or HIP are used,
> > as this ensures nothing is going to inject RTP packets
> > into the stream; using RTP can't ensure that.
> > 
> > 
> > (Crypto experts will point out there is a mafia fraud 
> > attack problem with the ICE technique as described 
> > in the paper.  That is true, but if you're not going 
> > to run SRTP using a secure keying mechanism you're 
> > suffering the same risk of someone injecting RTP into
> > your stream as the mafia fraud attack.)
> > 
> > -d
> > 


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> > >     <--- Enterprise-A ---->
> > >     clients --- PBX --- SBC --- SP1-SBC-1 --- SP1-SBC2 ----+
> > >                                                            |
> > >     +------------------------------------------------------+
> > >     |
> > >     |                              <--- Enterprise-B ---->
> > >     +--- SP2-SBC1 --- SP2-SBC2 --- SBC --- PBX --- clients
> > 
> > True.  I'll figure out a clearer way to diagram that.
> 
> Possibly it gets better if done up-down instead of left-right.

I'll fire up Jave and draw circles.

> > > Section 3, "mailto URLs" para:
> > > 
> > > sp1.net and sp2.net (and, later in the document, sp3.net) 
> are (or at
> > > least could be) real domains that shouldn't be used.
> > 
> > Agreed; I may go with "sp1.example.net" and "sp2.example.net", 
> > although this is less clear than using unique top-level domain 
> > names.
> 
> Perhaps it would be clearer to use for instance sp1.example.com,
> sp2.example.net and sp3.example.org.

That would work; thanks for the suggestion.

> > > Section 4.1, item about Contact:
> > > 
> > > How can the Contact header usefully be used in the 
> signing process? An
> > > SBC along the message path will happily replace it.
> > 
> > I have removed that in -01 (which isn't yet published, of 
> course).  There
> > was some thought it was necessary, but I agree it should be removed
> > from the signature.
> > 
> > On a similar note, I am considering removing CallId from 
> the signature.
> > Oftentimes the Call Id value contains an IP address (in 
> dotted decimal
> > or hex), and an SBC or B2BUA may also want to rewrite such 
> a CallId.  I
> > have made a note of that in -01 so this can be discussed.
> 
> I thought about that as well, but apparently I forgot about 
> adding it to
> my mail. I think it would be a good idea to take Call-ID out as well.

Ok, thanks.

-d

> Hans
> 
> -- 
> Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
> Ingate Systems AB  +46 13 210877   http://www.ingate.com/
> 
> Private: <unicorn@lysator.liu.se>  
> http://www.lysator.liu.se/~unicorn/


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> > Yes, I agree that will give the best chance.  Of course, it is
> > always possible for an SBC (or firewall) to block such messages 
> > anyway (as has been discussed on the RTPSEC mailer).
> 
> But that would at least mean that the SBC would have to look 
> inside the media stream to see that not everything in it is 
> (S)RTP, instead of just looking at the source and destination. 

Many SBCs, and some firewalls, include "RTP inspection".  They drop
packets that aren't RTP packets (such as ICE packets, DTLS-SRTP
packets [draft-ietf-avt-dtls-srtp], and RTCP packets
[draft-ietf-avt-rtp-and-rtcp-mux].

> It may happen, of course, but at least it's less likely. When it does
> appear, it would require an SBC that can recognize the security
> interchange for what it is and allow it to pass even though it isn't
> media. I'm not sure if that can reasonably be done.

Unless or until another solution to providing Identity is proposed,
I think it'll be necessary.  Identity is important so 
that the industry can build reliable and useful whitelists, 
blacklists, and reputation systems -- which are all necessary for 
call routing and mitigating spam.

-d


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> -----Original Message-----
> From: Rachel Albright [mailto:ralbrigh@cisco.com] 
> Sent: Monday, July 02, 2007 1:03 PM
> To: ietf-ipr@ietf.org
> Cc: dwing@cisco.com
> Subject: IPR Statement RE: draft-wing-sip-identity-media-00.txt
> 
> Cisco is the owner of one or more pending unpublished patent 
> applications relating to the subject matter of "SIP Identity 
> using Media Path" <draft-wing-sip-identity-media-00.txt>. 
>  
> If technology in this document is included in a standard 
> adopted by IETF and any claims of any Cisco patents are 
> necessary for practicing the standard, any party will have 
> the right to use any such patent claims under reasonable, 
> non-discriminatory terms, with reciprocity, to implement and 
> fully comply with the standard.
>  
> The reasonable non-discriminatory terms are:
>  
> If this standard is adopted, Cisco will not assert any 
> patents owned or controlled by Cisco against any party for 
> making, using, selling, importing or offering for sale a 
> product that implements the standard, provided, however that 
> Cisco retains the right to assert its patents (including the 
> right to claim past royalties) against any party that asserts 
> a patent it owns or controls (either directly or indirectly) 
> against Cisco or any of Cisco's affiliates or successors in 
> title or against any products of Cisco or any products of any 
> of Cisco's affiliates either alone or in combination with 
> other products; and Cisco retains the right to assert its 
> patents against any product or portion thereof that is not 
> necessary for compliance with the standard.
>  
> Royalty-bearing licenses will be available to anyone who 
> prefers that option.
> 
> For information contact:
>  
> Dan Lang
> Director, Patents and Standards IPR
> Cisco Systems
> +1 408-526-6672
> standards-ipr@cisco.com


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From sip-bounces@ietf.org Wed Jul 04 20:08:54 2007
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[this is a response to ostensibly "substantive" comments wrt the sip-saml draft]

Marcos Dytz <marcos.dytz@gmail.com> said  (via Hannes):
 >
 > 1) I believe that it would be good to leave a bit the emphasis on
 > authentication and providing different uses of the protocol

Sebastian Felis said (via Hannes):
 >
 > In my point of view, the draft focuses only on trait-based
 > authorization and tries to deploy trait-based authorization using SAML.
 > It does not try to specify SAML over SIP, which I was expected.


I'm assuming that both of the above are essentially saying:

   Perhaps the SIP SAML Profile should be generalized so it may be applied
   to other use cases in a SIP context.

yes?


This spec is presently designed for a fairly narrow use case, derived from 
RFC4474 "SIP Identity", and RFC4484 "Trait-Based Authorization Requirements for 
SIP". This use case is summarized in section 4 of draft-ietf-sip-saml-02 like so..

   4.  Specification Scope

      The scope of this specification is:

      o  Specify a SIP profile of SAML -- aka a "SIP SAML profile" -- such
         that a subject's profile attributes, and their domain's
         certificate, can be conveyed to a relying party using SAML.  In
         doing so, satisfy the requirements outlined in [RFC4484], and
         compose with [RFC4474].


Which arguably leaves essential things unsaid, such as..

  - "using SAML" here means "using a SAML assertion", not using one of the
    existing SAML profiles. This is a _new_, SIP-specific, SAML profile.

  - we're _not_ trying to concoct a fully-general "authorization mechanism"
    as imagined in RFC4484. Rather, we're simply trying to provide a means
    to convey (some undefined (sub)set of) a subject's profile attributes
    in the specific context of "authenticated identity management" as defined
    in RFC4474. As such, -sip-saml-02, presently meets (only) a subset of
    the requirements in RFC4484. For example, Requirement 3 of Section 5 of
    RFC4484 stipulates that an authorization assertion must be deliverable
    to a SIP UAC either by reference or by value, and -sip-saml-02
    presently defines by-reference only.

  - enhancing -sip-saml-02, meaning the "SIP SAML Profile", to address
    additional use cases, and/or fully meet RFC4484's requirements,
    would add significant complexity to the specification.

So, if the WG were to decide that -sip-saml-02 ought to, at this time, address 
additional use cases, and/or fully meet RFC4484's requirements, then we need to 
do yet more significant work on it. Otherwise, it seems fairly close to being 
complete.

Thoughts?

=JeffH

ps:

Just to clarify on this comment..

 > It does not try to specify SAML over SIP, which I was expected.

This seems to me to be indicative of a misunderstanding of SAML, because the 
phrase "SAML over SIP" doesn't make much actual sense. SAML per se is not "just 
a concrete protocol" that can be simply layered above another protocol, as HTTP 
is layered over TCP. Rather, it's a _framework_, that must be profiled for 
particular contexts of use, e.g. SIP Authenticated Identity Management. 
Sometimes folks, when thinking of SAML, are thinking of just one of the 
existing SAML profiles (e.g. Web SSO) and imagining that to be "SAML" or "the 
SAML protocol", and further imagine said "protocol" somehow being layered over 
some other protocol. This line of thinking wrt SAML isn't workable. Rather one 
needs to think in terms of making use of applicable pieces of SAML's "abstract 
classes" (in the "SAML core" specification) and freshly concretely applying 
them to the target use case and protocols, thus crafting a new SAML profile(s) 
(and binding(s)) -- which is exactly what -sip-saml-02 does.

See also..

How to Study and Learn SAML
http://identitymeme.org/how-to-study-and-learn-saml/


---
end







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From sip-bounces@ietf.org Wed Jul 04 23:21:38 2007
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Hi Eric et al,

I would appreciate this draft for detailed analysis for
the flaws of using INFO.

But since we are touching some proprietary
implementations now, 
I don't think we can ever prohibit implementation of
such usage even a new RFC is published.

I guess people started to use INFO just because it was
the only seemingly suitable mid-dialg
SIP Request at that time. In fact, for proprietary
implementation, people can still use other requests
(introduced later e.g. MESSAGE) for the same purpose. Do
we need to explicit prohibiting using of MESSAGE for
such purpose, if similar implementations evolve in the
future?

People are using INFO to fulfill some real requirements,
if we belive INFO is not the right way, 
I would propose we start to find and standardize more
prefered solution instead of just prohibiting using of
sth.
We've already done sth for DTMF, right?

Instead of updating RFC 2976 just for this purpose,
maybe an informational document to give some guidline is
enough.


Regards.
Peili



> -----Original Message-----
> From: Internet-Drafts@ietf.org
[mailto:Internet-Drafts@ietf.org]
> Sent: 03 July 2007 19:15
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-burger-sip-info-00.txt
> 
> A New Internet-Draft is available from the on-line
Internet-Drafts 
> directories.
> 
> 
> 	Title		: Session Initiation Protocol
(SIP) 
> INFO Method Context
> 	Author(s)	: E. Burger
> 	Filename	: draft-burger-sip-info-00.txt
> 	Pages		: 8
> 	Date		: 2007-7-3
> 	
>    The purpose of the INFO request for the Session
Initiation Protocol
>    (SIP), as described by RFC 2976, is to provide
mid-session SIP User
>    Agent (UA)-to-SIP UA application data transport.
In the years 
> since
>    the introduction of the INFO request, experience
with the use of 
> the
>    INFO request indicates a number of problems.  This
document 
> explains
>    why there   flaws of using INFO; and explains why
it is 
> not possible to create a
>    framework to rescue INFO for general purpose use.
Thus, this
>    document restricts the use of INFO to that
described in RFC 3372
>    (SIP-T).
>  are INFO-based, proprietary protocols in the wild;
the
> 
> 
> A URL for this Internet-Draft is:
>
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fre 2007-06-29 klockan 09:42 -0500 skrev Dean Willis:

> I'm pleased to announce that the authors think that the SIPS draft is  
> ready for working group last call. This draft has had the privilege  
> of more review prior to WGLC than most of our other efforts, so  
> perhaps it won't be too controversial.
> 
> Please make a final read through the draft and bring any comments  
> back to this list.


I've gone through the diff between the current version and the last one
I read, and on the whole it looks good to me. 

Some editorial nits are listed below.


Section 3.1.1, para 1:

> the Proxy is the TLS server side.  This directionality implies that

"proxy"

> directions (e..g, an incoming call), the UA must keep the TLS

"e.g."


Section 3.1.1, para 2:

> connection also solves the NAT and Firewall problem as it ensures

"firewall"


Section 3.1.2, para 1:

> place, the UAC would therefore take on the role of the TLS Client and

"client"

> SIP, a UA or a Proxy act both as UAC and UAS depending on if they are

"proxy"

> is very convenient.  This allows for TLS connections to be set-up or

"set up"


Section 3.1.2, para 3:

> most residential SIP deployments for example.  Mutual TLS can be used

Insert comma after "deployments".

> in those environments, but only if the connection is always stated by

"stated" is a strange word here. Do you mean "started"?


Section 3.1.2, last para:

> For this reasons, mutual TLS is mostly used in server-to-server

"this" -> "these"


Section 3.1.3, para 4:

> message (From, To, Request-URI, Contact header field, Route, etc.).

Why "Contact header field" and not just "Contact" like for the others?


Section 3.4, para 6:

> allowed, which would preclude the use of NATs or Firewalls.

"firewalls"


Section 4.1, para 3:

> SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed).  UAS that
>   do not support the SIPS URI scheme altogether "SHOULD reject the

"UAS" would be more readable as "UASs".

Change "altogether" to "at all".


Section 4.1, last para:

> response code 419 (SIPS Required)

Missing period at end of sentence.


Section 4.1.2, para 8:

> reject requests using the SIP Scheme with 419 (SIPS Required).  A UA

"scheme"


Section 4.2, para 5:

> response, it MUST NOT recurse on it.  The Proxy SHOULD forward the

"proxy"


Section 4.3, para 1:

> Using a Redirect Server with TLS instead of using a Proxy has some

"redirect server", "proxy"

> established connection between the Proxy and the UAS (such as with

"proxy"


Section A.3:

> (similar to what the transport=tls used to mean) has been suggested

Remove "the".


Hans

-- 
Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
Ingate Systems AB  +46 13 210857   http://www.ingate.com/

Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/


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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-outbound-09.txt 
Date: Thu, 5 Jul 2007 11:56:27 +0200
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(As WG chair)

We now have a new version of outbound. There hasn't exactly been a
flurry of comments since it was posted - does that mean it is now ready
for WGLC.

It has agenda time at the next IETF meeting, with a view to starting
WGLC not long after that meeting. If you believe there are issues with
the current version of this document, then you need to post them to the
list, so that they can be either resolved there, or addressed in the
meeting.

I am posting the extract of the minutes of the last meeting so that you
can remind yourself of the issues addressed in this new version.=20

Remember you can also gain access to diff versions here:

http://tools.ietf.org/html/draft-ietf-sip-outbound-09

Regards

Keith

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

Topic: Outbound
Led by Cullen Jennings
Slides presented and included in proceedings


Issue:  Keepalive for TCP, CRLF vs. Inline STUN
Extended discussion of this issue occurred, with each of several
factions represented.=20

It was noted that the current document dropped out discussion of using
tags in Path header to resolve configuration questions when Path is
used. This is to be repaired in the next revision of the document.

Following discussion the chairs polled on the question of whether the
room felt that the issue needed to be resolved "today" and whether they
would be willing to accept either approach based on a rough consensus.
There was a clear consensus to do so.

The chairs then polled the room on option 1 (STUN) vs option 2 (CRLF).
An approximate 2/3 majority favored CRLF.  The ADs decalred "rough
consensus" in favor of CRLF. This consensus is to be re-reviewed on the
mailing list.

Issue: When to do Keepalives?
Proposed that there is no longer a need for a keep-tcp parameter, and
that the keepalive-time parameter will suffice. There was no objection
noted to this proposal.

Issue: Simplifying Instance-ID Types
Proposed that SIPS-dependent option be dropped. No objections noted to
proprosal.

Issue: OPTIONS Probing for STUN support
Extended discussion occurred.

Proposed that discussion of using OPTIONS be deleted. No objections were
noted (just slight whimpering from one of the chairs).

Issue: Procedures for failures within 120s of REGISTER

Proposed that discussion be deleted. A consensus in favor of the
proposal was recorded.

Noted that Outbound will be tested at the next SIPIT, and that WGLC for
Outbound is imminent, so we should get any issues raised on the list
ASAP.

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

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Wednesday, June 27, 2007 8:50 PM
> To: i-d-announce@ietf.org
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-outbound-09.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> This draft is a work item of the Session Initiation Protocol=20
> Working Group of the IETF.
>=20
> 	Title		: Managing Client Initiated Connections=20
> in the Session Initiation Protocol (SIP)
> 	Author(s)	: C. Jennings, R. Mahy
> 	Filename	: draft-ietf-sip-outbound-09.txt
> 	Pages		: 42
> 	Date		: 2007-6-27
> =09
> The Session Initiation Protocol (SIP) allows proxy servers to
>    initiate TCP connections and send asynchronous UDP=20
> datagrams to User
>    Agents in order to deliver requests.  However, many practical
>    considerations, such as the existence of firewalls and Network
>    Address Translators (NATs), prevent servers from connecting to User
>    Agents in this way.  This specification defines behaviors for User
>    Agents, registrars and proxy servers that allow requests to be
>    delivered on existing connections established by the User=20
> Agent.  It
>    also defines keep alive behaviors needed to keep NAT bindings open
>    and specifies the usage of multiple connections.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
>=20
> To remove yourself from the I-D Announcement list, send a=20
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>=20
> Internet-Drafts are also available by anonymous FTP. Login=20
> with the username "anonymous" and a password of your e-mail=20
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>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html or=20
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-outbound-09.txt".
> =09
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
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> Below is the data which will enable a MIME compliant mail=20
> reader implementation to automatically retrieve the ASCII=20
> version of the Internet-Draft.
>=20


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DRAGE, Keith \(Keith\) writes:

 > We now have a new version of outbound. There hasn't exactly been a
 > flurry of comments since it was posted - does that mean it is now ready
 > for WGLC.

there was comment on it by someone from nokia, which made sense.  i
don't see a point of making a last call, before those issues have been
dealt with.

-- juha


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Agreed.

If there had been no comments I would have said that.

What I am trying to do at the moment is persuade all you people out
there to get stuff on the list (and start discussing it) before the IETF
meeting so we can address those issues before or in the IETF meeting.

If necessary, there will then be a revised draft, and we will WGLC that.

Regards

Keith

=20

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]=20
> Sent: Thursday, July 05, 2007 11:16 AM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-outbound-09.txt=20
>=20
> DRAGE, Keith \(Keith\) writes:
>=20
>  > We now have a new version of outbound. There hasn't=20
> exactly been a  > flurry of comments since it was posted -=20
> does that mean it is now ready  > for WGLC.
>=20
> there was comment on it by someone from nokia, which made=20
> sense.  i don't see a point of making a last call, before=20
> those issues have been dealt with.
>=20
> -- juha
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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DRAGE, Keith (Keith) wrote:
> (As WG chair)
> 
> We now have a new version of outbound. There hasn't exactly been a
> flurry of comments since it was posted - does that mean it is now ready
> for WGLC.
> 
> It has agenda time at the next IETF meeting, with a view to starting
> WGLC not long after that meeting. If you believe there are issues with
> the current version of this document, then you need to post them to the
> list, so that they can be either resolved there, or addressed in the
> meeting.

I never received an explanation of what kind of attacks could be avoided 
by an edge proxy using an algorithm that generates flow tokens that 
"cannot be modified by attackers" (section 5.2 of outbound-09).

This was the last e-mail in the thread the last time I brought it up on 
the list :

http://www1.ietf.org/mail-archive/web/sip/current/msg18336.html.

This leads me to think that either

   1) the security threats haven't been addressed properly

or

   2) that there in fact was no security threat so the requirement for an
      edge proxy to generate flows that "cannot be modified by attackers"
      is just unnecessary, and should therefor be removed
or

   3) there really _is_ something that gets significantly better by this
      requirement, and it should simply be explained in the draft.

/Fredrik


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Subject: Re: [Sip] Security issues with ietf-sip-loop-fix
Date: Thu, 5 Jul 2007 08:52:05 -0500
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To put a fine point on this - the kind of accidental loop you are  
talking about here
is one that does not have a fork in it.

It is possible (but much more unlikely) to set up something similar  
to the attack
in the draft by accident and a max-forwards as low as even 20 won't  
help much.

RjS

On Jul 4, 2007, at 6:07 AM, DRAGE, Keith ((Keith)) wrote:

> (as individual)
>
> With regard to:
>
>> If we go forward with path 1, the value of this draft is
>> probably not super high because Max-Forwards deals with the
>> the accidental loops in most cases  (but not all cases). If
>> we go with path 2, the WG will need make sure it OK with the
>> solution. I'm certainly open to other paths forward if
>> someone has a good idea.
>
> There have been comments previously on this list that the default  
> value
> of 70 for Max-Forwards is too high, and that a lower default value
> should have been used. Therefore Max-Forwards will find the accidental
> loops, but will take longer to do it, if the default values are used.
>
> Regards
>
> Keith
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Tuesday, July 03, 2007 3:36 AM
>> To: SIP
>> Subject: [Sip] Security issues with ietf-sip-loop-fix
>>
>>
>> (this email with my AD hat on)
>>
>> The type of SIP loops described in ietf-sip-loop-fix can be
>> roughly classified into two categories. Loops that had
>> happened because an accidental registration that causes a
>> loop, and loops that happened due to malicious intent.
>>
>> This draft attempted to deals with both. The abstract describes it as
>>
>>    This document normatively updates RFC 3261, the Session Initiation
>>     Protocol (SIP), to address a security vulnerability
>> identified in SIP
>>     proxy behavior.  This vulnerability enables an attack against SIP
>>     networks where a small number of legitimate, even authorized, SIP
>>     requests can stimulate massive amounts of proxy-to-proxy traffic.
>>
>> During IETF LC, substantial comment came up about the
>> security of this draft not really working in the case of
>> malicious loops. A thread ensues that convinced many people
>> on the thread that this draft did not really help with the
>> malicious loops. (Not 100% sure here but I suspect that the
>> authors and chairs were all convinced of this). I will let
>> others explain the issues that came up as we decide what to do next.
>>
>> I think that some of issue was discussed while the working
>> group was considering this draft - however, I suspect that
>> many participants did not understand how easily this security
>> would be defeated. I for one significantly underestimated the
>> how easy it was for a malicious attacker to cause this
>> problem. I apologize to the folks that attempted to explain
>> it to me and I just failed to "get it".
>>
>> Anyways, we need to figure out what to do next with this
>> draft. There are basically two paths forward that I see:
>>
>> 1) reduce the scope so that it only deals with accidental
>> loops and provides no protection against malicious loops. I
>> don't think this would change any of the mechanisms in the
>> draft but it would probably change lots of the text around
>> what security claims the draft made.
>>
>> 2) try to extend or replace the mechanism with one that does
>> protect against malicious attack. A strong candidate to
>> consider would be the work in draft-sparks-sipping-max-breadth.
>>
>> If we go forward with path 1, the value of this draft is
>> probably not super high because Max-Forwards deals with the
>> the accidental loops in most cases  (but not all cases). If
>> we go with path 2, the WG will need make sure it OK with the
>> solution. I'm certainly open to other paths forward if
>> someone has a good idea.
>>
>> What I would like to do now is start some discussion about
>> what the working group would like to do with this draft.
>>
>> Thanks, Cullen
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>
> _______________________________________________
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From: "Marcos Dytz" <marcos.dytz@gmail.com>
To: "Jeff Hodges" <Jeff.Hodges@neustar.biz>
Subject: Re: [Sip] SIP Saml Review by Marcos Dytz and Sebastian Felis
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Hello,

I agree on the limitations described below by Jeff. My concern was with the
emphasis on authentication throughout the draft, but I had in mind that it
was actually targeting that particular point (in the same way that
sip-paying targets payment).

Nevertheless, are there ideas to develop new drafts on further cases for
SIP/SAML? In my personal opinion, the protocol offers many technical
advantages (and flexibilities) that could be required for more general use
cases. By adopting and developing new cases, the overall protocol would only
be gaining a larger market share and could very well adopt some synergy
among the different areas.

As for Sebastian, I have the impression that he would like to see more
integration and deployment details. I had problems to fulfill the gaps on
integration while working on my thesis (e.g., enveloping SAML assertions in
SIP message - usually workflows target the transaction state and avoid
mentioning how to converge that with SIP - I did my implementation in the
way I thought best, but every developer is free to choose different formats
if specification does not define these little details).

As for your final points, agree to limitations due to complexity and partial
fulfilment of RFC4484. If further work is to be applied on the draft, I
believe the best action would be to leave the draft as it is and start
another work from scratch.

Regards,

Marcos

PS: I am rusty on software architecture. Can we describe SAML as a
framework? Isn't there a better terminology?

On 7/4/07, Jeff Hodges <Jeff.Hodges@neustar.biz> wrote:
>
> [this is a response to ostensibly "substantive" comments wrt the sip-saml
> draft]
>
> Marcos Dytz <marcos.dytz@gmail.com> said  (via Hannes):
> >
> > 1) I believe that it would be good to leave a bit the emphasis on
> > authentication and providing different uses of the protocol
>
> Sebastian Felis said (via Hannes):
> >
> > In my point of view, the draft focuses only on trait-based
> > authorization and tries to deploy trait-based authorization using SAML.
> > It does not try to specify SAML over SIP, which I was expected.
>
>
> I'm assuming that both of the above are essentially saying:
>
>   Perhaps the SIP SAML Profile should be generalized so it may be applied
>   to other use cases in a SIP context.
>
> yes?
>
>
> This spec is presently designed for a fairly narrow use case, derived from
> RFC4474 "SIP Identity", and RFC4484 "Trait-Based Authorization
> Requirements for
> SIP". This use case is summarized in section 4 of draft-ietf-sip-saml-02
> like so..
>
>   4.  Specification Scope
>
>      The scope of this specification is:
>
>      o  Specify a SIP profile of SAML -- aka a "SIP SAML profile" -- such
>         that a subject's profile attributes, and their domain's
>         certificate, can be conveyed to a relying party using SAML.  In
>         doing so, satisfy the requirements outlined in [RFC4484], and
>         compose with [RFC4474].
>
>
> Which arguably leaves essential things unsaid, such as..
>
> - "using SAML" here means "using a SAML assertion", not using one of the
>    existing SAML profiles. This is a _new_, SIP-specific, SAML profile.
>
> - we're _not_ trying to concoct a fully-general "authorization mechanism"
>    as imagined in RFC4484. Rather, we're simply trying to provide a means
>    to convey (some undefined (sub)set of) a subject's profile attributes
>    in the specific context of "authenticated identity management" as
> defined
>    in RFC4474. As such, -sip-saml-02, presently meets (only) a subset of
>    the requirements in RFC4484. For example, Requirement 3 of Section 5 of
>    RFC4484 stipulates that an authorization assertion must be deliverable
>    to a SIP UAC either by reference or by value, and -sip-saml-02
>    presently defines by-reference only.
>
> - enhancing -sip-saml-02, meaning the "SIP SAML Profile", to address
>    additional use cases, and/or fully meet RFC4484's requirements,
>    would add significant complexity to the specification.
>
> So, if the WG were to decide that -sip-saml-02 ought to, at this time,
> address
> additional use cases, and/or fully meet RFC4484's requirements, then we
> need to
> do yet more significant work on it. Otherwise, it seems fairly close to
> being
> complete.
>
> Thoughts?
>
> =JeffH
>
> ps:
>
> Just to clarify on this comment..
>
> > It does not try to specify SAML over SIP, which I was expected.
>
> This seems to me to be indicative of a misunderstanding of SAML, because
> the
> phrase "SAML over SIP" doesn't make much actual sense. SAML per se is not
> "just
> a concrete protocol" that can be simply layered above another protocol, as
> HTTP
> is layered over TCP. Rather, it's a _framework_, that must be profiled for
> particular contexts of use, e.g. SIP Authenticated Identity Management.
> Sometimes folks, when thinking of SAML, are thinking of just one of the
> existing SAML profiles (e.g. Web SSO) and imagining that to be "SAML" or
> "the
> SAML protocol", and further imagine said "protocol" somehow being layered
> over
> some other protocol. This line of thinking wrt SAML isn't workable. Rather
> one
> needs to think in terms of making use of applicable pieces of SAML's
> "abstract
> classes" (in the "SAML core" specification) and freshly concretely
> applying
> them to the target use case and protocols, thus crafting a new SAML
> profile(s)
> (and binding(s)) -- which is exactly what -sip-saml-02 does.
>
> See also..
>
> How to Study and Learn SAML
> http://identitymeme.org/how-to-study-and-learn-saml/
>
>
> ---
> end
>
>
>
>
>
>

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<div>Hello,</div>
<div>&nbsp;</div>
<div>I agree on the limitations described below by Jeff. My concern was with the emphasis on authentication throughout the draft, but I had in mind that it was actually targeting that particular point (in the same way that sip-paying targets payment).
</div>
<div>&nbsp;</div>
<div>Nevertheless, are there ideas to develop new drafts on further cases for SIP/SAML? In my personal opinion, the protocol offers&nbsp;many technical advantages (and flexibilities) that could be&nbsp;required&nbsp;for&nbsp;more general use cases. By adopting and developing new cases, the overall protocol would only be gaining a larger market share and could very well adopt some&nbsp;synergy among the different areas.
<br>&nbsp;</div>
<div>As for Sebastian, I have the impression that he would like to see more integration and deployment details. I had problems to fulfill the gaps on integration while working on my thesis (e.g., enveloping SAML assertions in SIP message - usually workflows target the transaction state and avoid mentioning how to&nbsp;converge that&nbsp;with SIP - I did&nbsp;my implementation&nbsp;in the way I thought best, but every developer is free to choose different formats if specification does not define these little details).
</div>
<div>&nbsp;</div>
<div>As for your final points, agree to limitations due to complexity and partial fulfilment of RFC4484. If further work is to be applied on the draft, I believe the best action would be to leave the draft as it is and start another work from scratch.
</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Marcos</div>
<div>&nbsp;</div>
<div>PS: I am rusty on software architecture. Can we describe SAML as a framework? Isn&#39;t there a better terminology?<br>&nbsp;</div>
<div><span class="gmail_quote">On 7/4/07, <b class="gmail_sendername">Jeff Hodges</b> &lt;<a href="mailto:Jeff.Hodges@neustar.biz">Jeff.Hodges@neustar.biz</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">[this is a response to ostensibly &quot;substantive&quot; comments wrt the sip-saml draft]<br><br>Marcos Dytz &lt;
<a href="mailto:marcos.dytz@gmail.com">marcos.dytz@gmail.com</a>&gt; said&nbsp;&nbsp;(via Hannes):<br>&gt;<br>&gt; 1) I believe that it would be good to leave a bit the emphasis on<br>&gt; authentication and providing different uses of the protocol
<br><br>Sebastian Felis said (via Hannes):<br>&gt;<br>&gt; In my point of view, the draft focuses only on trait-based<br>&gt; authorization and tries to deploy trait-based authorization using SAML.<br>&gt; It does not try to specify SAML over SIP, which I was expected.
<br><br><br>I&#39;m assuming that both of the above are essentially saying:<br><br>&nbsp;&nbsp;Perhaps the SIP SAML Profile should be generalized so it may be applied<br>&nbsp;&nbsp;to other use cases in a SIP context.<br><br>yes?<br><br><br>
This spec is presently designed for a fairly narrow use case, derived from<br>RFC4474 &quot;SIP Identity&quot;, and RFC4484 &quot;Trait-Based Authorization Requirements for<br>SIP&quot;. This use case is summarized in section 4 of draft-ietf-sip-saml-02 like so..
<br><br>&nbsp;&nbsp;4.&nbsp;&nbsp;Specification Scope<br><br>&nbsp;&nbsp;&nbsp;&nbsp; The scope of this specification is:<br><br>&nbsp;&nbsp;&nbsp;&nbsp; o&nbsp;&nbsp;Specify a SIP profile of SAML -- aka a &quot;SIP SAML profile&quot; -- such<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;that a subject&#39;s profile attributes, and their domain&#39;s
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;certificate, can be conveyed to a relying party using SAML.&nbsp;&nbsp;In<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;doing so, satisfy the requirements outlined in [RFC4484], and<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;compose with [RFC4474].<br><br><br>Which arguably leaves essential things unsaid, such as..
<br><br>- &quot;using SAML&quot; here means &quot;using a SAML assertion&quot;, not using one of the<br>&nbsp;&nbsp; existing SAML profiles. This is a _new_, SIP-specific, SAML profile.<br><br>- we&#39;re _not_ trying to concoct a fully-general &quot;authorization mechanism&quot;
<br>&nbsp;&nbsp; as imagined in RFC4484. Rather, we&#39;re simply trying to provide a means<br>&nbsp;&nbsp; to convey (some undefined (sub)set of) a subject&#39;s profile attributes<br>&nbsp;&nbsp; in the specific context of &quot;authenticated identity management&quot; as defined
<br>&nbsp;&nbsp; in RFC4474. As such, -sip-saml-02, presently meets (only) a subset of<br>&nbsp;&nbsp; the requirements in RFC4484. For example, Requirement 3 of Section 5 of<br>&nbsp;&nbsp; RFC4484 stipulates that an authorization assertion must be deliverable
<br>&nbsp;&nbsp; to a SIP UAC either by reference or by value, and -sip-saml-02<br>&nbsp;&nbsp; presently defines by-reference only.<br><br>- enhancing -sip-saml-02, meaning the &quot;SIP SAML Profile&quot;, to address<br>&nbsp;&nbsp; additional use cases, and/or fully meet RFC4484&#39;s requirements,
<br>&nbsp;&nbsp; would add significant complexity to the specification.<br><br>So, if the WG were to decide that -sip-saml-02 ought to, at this time, address<br>additional use cases, and/or fully meet RFC4484&#39;s requirements, then we need to
<br>do yet more significant work on it. Otherwise, it seems fairly close to being<br>complete.<br><br>Thoughts?<br><br>=JeffH<br><br>ps:<br><br>Just to clarify on this comment..<br><br>&gt; It does not try to specify SAML over SIP, which I was expected.
<br><br>This seems to me to be indicative of a misunderstanding of SAML, because the<br>phrase &quot;SAML over SIP&quot; doesn&#39;t make much actual sense. SAML per se is not &quot;just<br>a concrete protocol&quot; that can be simply layered above another protocol, as HTTP
<br>is layered over TCP. Rather, it&#39;s a _framework_, that must be profiled for<br>particular contexts of use, e.g. SIP Authenticated Identity Management.<br>Sometimes folks, when thinking of SAML, are thinking of just one of the
<br>existing SAML profiles (e.g. Web SSO) and imagining that to be &quot;SAML&quot; or &quot;the<br>SAML protocol&quot;, and further imagine said &quot;protocol&quot; somehow being layered over<br>some other protocol. This line of thinking wrt SAML isn&#39;t workable. Rather one
<br>needs to think in terms of making use of applicable pieces of SAML&#39;s &quot;abstract<br>classes&quot; (in the &quot;SAML core&quot; specification) and freshly concretely applying<br>them to the target use case and protocols, thus crafting a new SAML profile(s)
<br>(and binding(s)) -- which is exactly what -sip-saml-02 does.<br><br>See also..<br><br>How to Study and Learn SAML<br><a href="http://identitymeme.org/how-to-study-and-learn-saml/">http://identitymeme.org/how-to-study-and-learn-saml/
</a><br><br><br>---<br>end<br><br><br><br><br><br></blockquote></div><br>

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From sip-bounces@ietf.org Thu Jul 05 11:48:33 2007
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From: Byron Campen <bcampen@estacado.net>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-09.txt
Date: Thu, 5 Jul 2007 10:47:06 -0500
To: Fredrik Thulin <ft@it.su.se>
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	I am in agreement here. There are plenty of ways to hijack a  
registration without forging a flow-token. We could just remove the  
entire Record-Route/Path stack (depending on whether we're hijacking  
a dialog, or a registration), and substitute one of our own,  
bypassing any logic at the proxies entirely. If the network is set up  
in such a way that doing so would prevent requests from being  
successfully forwarded, we could always remove the flow-token from  
the last hop, and insert a Record-Route/Path header field value that  
would cause stuff to be routed anywhere we please.

	I suppose if the edge-proxy _only_ accepts Route headers with flow  
tokens (ie, Route headers without will just cause the request to  
fail), and bypassing the edge-proxy is impossible (ie, the network,  
including endpoints, is configured to forward all traffic to the edge  
proxy regardless of what is in the Route headers), cryptographic  
protection of flow tokens would help. But, in the general case, it  
will not afford us any meaningful protection. We need to rely on  
something like TLS (which, conveniently enough, outbound makes it  
easy to do).

Best regards,
Byron Campen


> DRAGE, Keith (Keith) wrote:
>> (As WG chair)
>> We now have a new version of outbound. There hasn't exactly been a
>> flurry of comments since it was posted - does that mean it is now  
>> ready
>> for WGLC.
>> It has agenda time at the next IETF meeting, with a view to starting
>> WGLC not long after that meeting. If you believe there are issues  
>> with
>> the current version of this document, then you need to post them  
>> to the
>> list, so that they can be either resolved there, or addressed in the
>> meeting.
>
> I never received an explanation of what kind of attacks could be  
> avoided by an edge proxy using an algorithm that generates flow  
> tokens that "cannot be modified by attackers" (section 5.2 of  
> outbound-09).
>
> This was the last e-mail in the thread the last time I brought it  
> up on the list :
>
> http://www1.ietf.org/mail-archive/web/sip/current/msg18336.html.
>
> This leads me to think that either
>
>   1) the security threats haven't been addressed properly
>
> or
>
>   2) that there in fact was no security threat so the requirement  
> for an
>      edge proxy to generate flows that "cannot be modified by  
> attackers"
>      is just unnecessary, and should therefor be removed
> or
>
>   3) there really _is_ something that gets significantly better by  
> this
>      requirement, and it should simply be explained in the draft.
>
> /Fredrik
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From sip-bounces@ietf.org Thu Jul 05 16:18:46 2007
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Subject: [Sip] Open issue in the hitchhikers guide - seriously! Relation to
 essential corrections
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I was working on my regular update to the sip hitchhikers guide:
http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-02.txt

and ran into an issue. We are now pursuing the SIP essential corrections 
process:
http://www.ietf.org/internet-drafts/draft-drage-sip-essential-correction-01.txt

which has begun to produce internet drafts, that are draft-ietf-sip-XXX, 
which will be tracked and produced by the working group. The question 
is, do these drafts get included into the hitchhikers guide or not. My 
view on this, is NO, they do not. The reason is that these drafts will 
never reach RFC, only the collected essential correction RFC. So, once 
that first RFC is produced, the hitchhikers guide would reference that, 
and that would encompass everything.

So, in the interim, what I'd propose is that the running I-D version of 
hitchhikers guide have an "essential corrections" draft, which in fact 
references all of the I-Ds currently in line for essential correction. 
Unfortunately, its not clear which drafts those are. record-route-fix, 
yes. What about loop-fix? Is that going in? It would be convenient to 
have a way of clearly knowing what drafts are going to be for essential 
corrections. Is the rule:

draft-ietf-sip-*-fix

are drafts targeted for essential correction? If so, I think that should 
be clearly stated in the drage-essential-corrections draft.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org Thu Jul 05 16:50:05 2007
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Subject: Re: [Sip] RE: I-D ACTION:draft-burger-sip-info-00.txt
From: Eric Burger <eburger@bea.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Good catch.  I'll fix that if there is another draft.


On 7/4/07 9:31 AM, "Elwell, John" <john.elwell@siemens.com> wrote:

> Eric,
> 
> I would just like to correct a slight inaccuracy in this draft. RFC 3372
> does not specify the use of INFO for QSIG, but RFC 3204 (which is
> referenced by RFC 3372) does define the MIME type for QSIG. The
> transport of QSIG using the SIP INFO method is instead specified in
> Standard ECMA-355 from Ecma-International. This is indeed a very similar
> usage of INFO to that specified in RFC 3372 and should be allowed in the
> proposed update to RFC 2976.
> 
> John 
> 
>> -----Original Message-----
>> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
>> Sent: 03 July 2007 19:15
>> To: i-d-announce@ietf.org
>> Subject: I-D ACTION:draft-burger-sip-info-00.txt
>> 
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>> 
>> 
>> Title  : Session Initiation Protocol (SIP)
>> INFO Method Context
>> Author(s) : E. Burger
>> Filename : draft-burger-sip-info-00.txt
>> Pages  : 8
>> Date  : 2007-7-3
>> 
>>    The purpose of the INFO request for the Session Initiation Protocol
>>    (SIP), as described by RFC 2976, is to provide mid-session SIP User
>>    Agent (UA)-to-SIP UA application data transport.  In the
>> years since
>>    the introduction of the INFO request, experience with the
>> use of the
>>    INFO request indicates a number of problems.  This
>> document explains
>>    why there   flaws of using INFO; and explains why it is
>> not possible to create a
>>    framework to rescue INFO for general purpose use.  Thus, this
>>    document restricts the use of INFO to that described in RFC 3372
>>    (SIP-T).
>>  are INFO-based, proprietary protocols in the wild; the
>> 
>> 
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-burger-sip-info-00.txt
>> 
>> To remove yourself from the I-D Announcement list, send a message to
>> i-d-announce-request@ietf.org with the word unsubscribe in
>> the body of 
>> the message. 
>> You can also visit
>> https://www1.ietf.org/mailman/listinfo/I-D-announce
>> to change your subscription settings.
>> 
>> Internet-Drafts are also available by anonymous FTP. Login with the
>> username "anonymous" and a password of your e-mail address. After
>> logging in, type "cd internet-drafts" and then
>> "get draft-burger-sip-info-00.txt".
>> 
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> 
>> Internet-Drafts can also be obtained by e-mail.
>> 
>> Send a message to:
>> mailserv@ietf.org.
>> In the body type:
>> "FILE /internet-drafts/draft-burger-sip-info-00.txt".
>> 
>> NOTE: The mail server at ietf.org can return the document in
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>> command.  To decode the response(s), you will need "munpack" or
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>> mail readers
>> exhibit different behavior, especially when dealing with
>> "multipart" MIME messages (i.e. documents which have been split
>> up into multiple messages), so check your local documentation on
>> how to manipulate these messages.
>> 
>> Below is the data which will enable a MIME compliant mail reader
>> implementation to automatically retrieve the ASCII version of the
>> Internet-Draft.
>> 
> 
> 
> _______________________________________________
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From sip-bounces@ietf.org Thu Jul 05 16:56:30 2007
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Subject: Re: [Sip] comments on draft-burger-sip-info-00.txt
From: Eric Burger <eburger@bea.com>
To: Peili Xu <xupeili@huawei.com>
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I was trying to be subtle for those that knew, and not so subtle for those
that do not folks.

Believe me, the folks at my old (and, for that matter, new) employer would
probably get upset if I said, "MSCML is not allowed to work."  Likewise, I
think folks at my old nemesis and my new found best friends would get upset
if I said, "MSML is not allowed to work."

However, I did NOT want to be seen as leaving the door open for more and
more uses of INFO.  What the draft sets out to do is say, "The one and only
IETF sanctioned use of INFO is for the SIP-T case."  I.e., there won't be
any Standards Track INFO RFC's.  However, to keep everyone with a legacy
happy, Section 4 subtly states:

   In recognition of existing, proprietary use of INFO, proxies MUST NOT
   take any action other than that described by RFC 3261 and RFC 2976
   with respect to handling INFO requests.

This is the escape hatch: one can still use INFO, subject to all of the
dangers enumerated in Section 2.  Moreover, the draft explicitly states that
SIP "network elements" must not "add value" by killing INFO messages they do
not understand.


On 7/4/07 11:20 PM, "Peili Xu" <xupeili@huawei.com> wrote:

> 
>  
> Hi Eric et al,
> 
> I would appreciate this draft for detailed analysis for
> the flaws of using INFO.
> 
> But since we are touching some proprietary
> implementations now,
> I don't think we can ever prohibit implementation of
> such usage even a new RFC is published.
> 
> I guess people started to use INFO just because it was
> the only seemingly suitable mid-dialg
> SIP Request at that time. In fact, for proprietary
> implementation, people can still use other requests
> (introduced later e.g. MESSAGE) for the same purpose. Do
> we need to explicit prohibiting using of MESSAGE for
> such purpose, if similar implementations evolve in the
> future?
> 
> People are using INFO to fulfill some real requirements,
> if we belive INFO is not the right way,
> I would propose we start to find and standardize more
> prefered solution instead of just prohibiting using of
> sth.
> We've already done sth for DTMF, right?
> 
> Instead of updating RFC 2976 just for this purpose,
> maybe an informational document to give some guidline is
> enough.
> 
> 
> Regards.
> Peili
> 
> 
> 
>> -----Original Message-----
>> From: Internet-Drafts@ietf.org
> [mailto:Internet-Drafts@ietf.org]
>> Sent: 03 July 2007 19:15
>> To: i-d-announce@ietf.org
>> Subject: I-D ACTION:draft-burger-sip-info-00.txt
>> 
>> A New Internet-Draft is available from the on-line
> Internet-Drafts 
>> directories.
>> 
>> 
>> Title  : Session Initiation Protocol
> (SIP) 
>> INFO Method Context
>> Author(s) : E. Burger
>> Filename : draft-burger-sip-info-00.txt
>> Pages  : 8
>> Date  : 2007-7-3
>> 
>>    The purpose of the INFO request for the Session
> Initiation Protocol
>>    (SIP), as described by RFC 2976, is to provide
> mid-session SIP User
>>    Agent (UA)-to-SIP UA application data transport.
> In the years 
>> since
>>    the introduction of the INFO request, experience
> with the use of 
>> the
>>    INFO request indicates a number of problems.  This
> document 
>> explains
>>    why there   flaws of using INFO; and explains why
> it is 
>> not possible to create a
>>    framework to rescue INFO for general purpose use.
> Thus, this
>>    document restricts the use of INFO to that
> described in RFC 3372
>>    (SIP-T).
>>  are INFO-based, proprietary protocols in the wild;
> the
>> 
>> 
>> A URL for this Internet-Draft is:
>> 
> http://www.ietf.org/internet-drafts/draft-burger-sip-inf
> o-00.txt
>> 
>> To remove yourself from the I-D Announcement list,
> send a message to
>> i-d-announce-request@ietf.org with the word
> unsubscribe in the body of
>> the message.
>> You can also visit
>> https://www1.ietf.org/mailman/listinfo/I-D-announce
>> to change your subscription settings.
>> 
>> Internet-Drafts are also available by anonymous FTP.
> Login with the 
>> username "anonymous" and a password of your e-mail
> address. After 
>> logging in, type "cd internet-drafts" and then "get
>> draft-burger-sip-info-00.txt".
>> 
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html or
>> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> 
>> Internet-Drafts can also be obtained by e-mail.
>> 
>> Send a message to:
>> mailserv@ietf.org.
>> In the body type:
>> "FILE
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>> 
>> NOTE: The mail server at ietf.org can return the
> document in
>> MIME-encoded form by using the "mpack" utility.
> To use this
>> feature, insert the command "ENCODING mime"
> before the "FILE"
>> command.  To decode the response(s), you will
> need "munpack" or
>> a MIME-compliant mail reader.  Different
> MIME-compliant mail readers
>> exhibit different behavior, especially when
> dealing with
>> "multipart" MIME messages (i.e. documents which
> have been split
>> up into multiple messages), so check your local
> documentation on
>> how to manipulate these messages.
>> 
>> Below is the data which will enable a MIME compliant
> mail reader 
>> implementation to automatically retrieve the ASCII
> version of the 
>> Internet-Draft.
>> 
> 
> 
> _______________________________________________
> Sip mailing list
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol Use sip-implementors@cs.columbia.edu for
> questions on current sip Use sipping@ietf.org for new
> developments on the application of sip
> 
> 
> 
> 
> _______________________________________________
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Hi,=20

>My concern with P-Called-Party-ID is similar to my concern on=20
>History-Info. On History-Info, the draft states:
>=20
>     Firstly, it would cause the Request-URI to be relegated to nothing
>     more than an indicator of the next hop for the request, identical
>     exactly to the purpose of the Route header field. This results in
>     two things in the SIP specification which do exactly the same =
thing.
>     Worse still, this is not just for some small feature of SIP (where
>     such duplication might not be a big deal), but rather, it would be =
a
>     duplication of SIP's primary function - routing of a call towards =
a
>     target.
>=20
> This concern applies exactly to P-Called-Party-ID. Indeed,=20
> its even worse, since now we would have, in a request:
>=20
> Request-URI
> To
> P-Called-Party-ID
> History-Info
> Route
>=20
>and I challenge you to explain to implementors what the=20
>differences are.=20
>What would happen when they are inconsistent?
>
>My objective was to reduce this count. Indeed I was hopeful=20
>that UA-loose-route would allow us to deprecate=20
>P-Called-Party-ID, not the other way around. WIth=20
>UA-loose-route, we end up with:
>=20
>Request-URI: identifies the target
>Route: tells you how to get to the target
>To: tells you the original target
>History info: sequence of identities of targets

If people can't explain to their implementors the meaning of different =
SIP headers maybe THAT is something we should concentrate on, and if =
needed write a draft about.

>which is much cleaner.

If we would re-do SIP I am sure lots of things could be done cleaner.

P-Called-Party-ID does not change the current routing mechanism, and it =
does not have the potential issue with proxies (and possible also SBCs) =
that you have mentioned. And the proxy re-writing the R-URI can use the =
mechanism no matter if the UA supports it or not.=20

(And, naturally the mechanism would of course be backward compatible =
with entities and networks that already support P-Called-Party-ID.)

>So, functionally, P-Called-Party-ID could work. Any number of things =
*can* work.

Yes, but P-Called-Party-ID is a standardized mechanism, made for this =
purpose, which most likely does work.

Regards,

Christer




> Christer Holmberg (JO/LMF) wrote:
> > Hi,
> >=20
> >> It is entirely appropriate at this stage to put forward=20
> alternatives.=20
> >> It is even more appropriate to do it now rather than in 6 months=20
> >> time.
> >>=20
> >> However please do not expect the editor to elaborate them.=20
> If the WG=20
> >> has ideas, send text, either by individual author drafts,=20
> or by the=20
> >> mailing list.
> >=20
> > Regarding the P-Called-Party header, there already IS an RFC=20
> > describing the header, and what it is used for. So, IF it already=20
> > solves the problem (RFC3455 does say that the header is=20
> used to convey=20
> > the AOR to the UAS), why would we need a separate draft? The only=20
> > additional information that a draft could contain are the use-cases=20
> > where receiving the AOR is important - but those use-cases are=20
> > solution independent and already described in Jonathan's draft.
> >=20
> > Of course, if we want to define a NEW header, or a NEW URI=20
> parameter,=20
> > that may be done in a separate draft.
> >=20
> > But, no matter how we do it, I think the author should at least=20
> > indicate whether he has already thought about the alternatives that=20
> > have been presented, and why he thinks that they don't work.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >>> -----Original Message----- From: Christer Holmberg (JO/LMF)=20
> >>> [mailto:christer.holmberg@ericsson.com] Sent: Thursday, June 14,
> >>> 2007 10:51 AM To: DRAGE, Keith (Keith); Jonathan=20
> Rosenberg; IETF SIP=20
> >>> List Subject: RE: [Sip] UA Loose Routing
> >>>=20
> >>>=20
> >>> Hi Keith,
> >>>=20
> >>> The text you sent says:
> >>>=20
> >>> "Consensus noted that we do need some sort of solution for
> >> the general
> >>> problem of delivery of URI parameters across contact-routing=20
> >>> operations."
> >>>=20
> >>> I have no problems with that.
> >>>=20
> >>> I also have no problem with using Jonathan's draft as base line, =20
> >>> because the draft discusses the problem in a good way.
> >>>=20
> >>> But, still: as the text says, the agreement was to come up
> >> with SOME
> >>> SORT OF SOLUTION, which to me does not mean that we have=20
> agreed to=20
> >>> adopt the specific solution in Jonathan's draft. We=20
> should also be=20
> >>> "allowed" to discuss alternative solutions, whether they
> >>>=20
> >> are based on
> >>> P-Called-Party, some other header, some URI parameter, or
> >> whatever...
> >>> Maybe those alternative solutions do not work, and then=20
> we should =20
> >>> describe that (as Jonathan has done for the To header-=20
> and History=20
> >>> header alternatives).
> >>>=20
> >>> Regards,
> >>>=20
> >>> Christer
> >>>=20
> >>>=20
> >>>=20
> >>>=20
> >>>> -----Original Message----- From: DRAGE, Keith (Keith)=20
> >>>> [mailto:drage@alcatel-lucent.com] Sent: 14. kes=E4kuuta 2007 =
12:40=20
> >>>> To: Christer Holmberg (JO/LMF); Jonathan Rosenberg; IETF=20
> SIP List=20
> >>>> Subject: RE: [Sip] UA Loose Routing
> >>>>=20
> >>>> You appear to be questioning whether the solution was
> >> adopted. The
> >>>> minutes state:
> >>>>=20
> >>>>=20
> >> -------------------------------------------------------------------
> >>=20
> >>>> Topic: GRUU and Supporting Drafts Led by Jonathan=20
> Rosenberg Slides=20
> >>>> included in proceedings Read:
> >>>> draft-ietf-sip-gruu-11,draft-rosenberg-sip-ua-loose-route-00
> >>>>=20
> >>>> Change since last version reviewed.
> >>>>=20
> >>>> Issue: GRUU subtype names. The room demonstrated a slight
> >>> preference
> >>>> for the names used in -11 over known alternatives, but
> >> nobody seems
> >>>> particularly enamored with these names and the floor
> >>> remains open to
> >>>> suggestions.
> >>>>=20
> >>>> Discussion on ua-loose-route draft deferred to next session.
> >>>>=20
> >>>>=20
> >>>> Session 2: Friday 0900-1130 Grand Ballroom C
> >>>>=20
> >>>>=20
> >>>> Topic: UA-loose-route, continued from previous session.
> >>>>=20
> >>>> Consensus noted that we do need some sort of solution for
> >>> the general
> >>>> problem of delivery of URI parameters across contact-routing=20
> >>>> operations.
> >>>>=20
> >>>> Issue: Concerns raised about backward compatibility. Noted that =20
> >>>> compliant P-CSCF should be OK.  Suggested that we have a
> >>> design team
> >>>> revisit call-flows and analyze for backward compatibility issues.
> >>>>=20
> >>>> Issue: Does GRUU normatively depend on this work? Resolved:=20
> >>>> GRUU does not reference this document. However, some
> >>> participants feel
> >>>> it should, as the benefit of GRUU cannot be fully
> >> recognized in the
> >>>> absence of this specification.  The chairs asked for a
> >> hum, and the
> >>>> consensus of the room is that GRUU and ua-loose-route=20
> can proceed=20
> >>>> independently.
> >>>>=20
> >>>> Question: Adopt this draft as baseline text for a WG
> >> effort, should
> >>>> the ADs approve the milestone? Consensus to adopt noted=20
> by chairs.=20
> >>>> To-do: Chairs to work with AD to set new milestone.
> >>>>=20
> >>=20
> ---------------------------------------------------------------------
> >> -
> >>=20
> >>>> The minutes quite clearly say "Adopt this draft as baseline
> >>> text for a
> >>>> WG effort" and given that the charter item now exists, then
> >>> I believe
> >>>> that we are entitled to call for the editor (Jonathan) to
> >>> submit the
> >>>> current text as draft-ietf-sip-ua-loose-route.
> >>>>=20
> >>>> I agree the file name says loose-route, but I don'y think
> >> we should
> >>>> change that. After all, we lose that filename when it gets
> >>> published
> >>>> as an RFC anyway.
> >>>>=20
> >>>> The charter item states
> >>>>=20
> >>>> Dec 2007    Delivering request-URI and parameters to UAS via=20
> >>>> proxy to WGLC Feb 2008    Delivering request-URI and parameters
> >>>> to UAS via proxy to IESG (PS)
> >>>>=20
> >>>> I assume that you are OK with this.
> >>>>=20
> >>>> What I would suggest is that you address specific=20
> comments to the=20
> >>>> title of the deliverable:
> >>>>=20
> >>>> Applying Loose Routing to Session Initiation Protocol (SIP) User=20
> >>>> Agents (UA)
> >>>>=20
> >>>> And to the abstract:
> >>>>=20
> >>>> Abstract
> >>>>=20
> >>>> A key part of the behavior of the Session Initiation
> >>> Protocol (SIP)
> >>>> is that SIP proxies rewrite the Request-URI as a request moves=20
> >>>> throughout the network.  Over the years, experience has
> >>> shown this
> >>>> to be problematic.  It makes it difficult to use Request URI for=20
> >>>> service invocation, complicates emergency services, makes it
> >>> more complex
> >>>> to support aliases, and so on.  Architecturally, it=20
> confounds the=20
> >>>> concepts of address and route.  This document proposes
> >> to change
> >>>> this through a new mechanism called UA loose routing.
> >>>>=20
> >>>> As you think appropriate to your concerns.
> >>>>=20
> >>>> Alternatively, if you (or indeed any other SIP WG member)
> >> do have a
> >>>> completely different solution, then there is still time
> >> to post an
> >>>> individual draft or post a skeleton of a mechanism to the list.
> >>>>=20
> >>>>=20
> >>>> Regards
> >>>>=20
> >>>> Keith
> >>>>=20
> >>>>> -----Original Message----- From: Christer Holmberg (JO/LMF)=20
> >>>>> [mailto:christer.holmberg@ericsson.com] Sent:=20
> Wednesday, June 13,=20
> >>>>> 2007 8:52 PM To: DRAGE, Keith (Keith); Jonathan Rosenberg; IETF=20
> >>>>> SIP List Subject: RE: [Sip] UA Loose Routing
> >>>>>=20
> >>>>>=20
> >>>>> Hi,
> >>>>>=20
> >>>>> We can call it loose route if we want, but I think the
> >>>> initial focus
> >>>>> should be the problem we are going to solve (I don't think
> >>>> we agreed
> >>>>> on a specific solution at #67), and we should look into
> >>> alternative
> >>>>> solutions (the draft does that in some extent, but
> >> other possible
> >>>>> alternatives have also been mentioned).
> >>>>>=20
> >>>>> Or, do you prefer to have separate drafts describing
> >> alternative
> >>>>> solutions for the problem?
> >>>>>=20
> >>>>> Regards,
> >>>>>=20
> >>>>> Christer
> >>>>>=20
> >>>>>=20
> >>>>>=20
> >>>>>> -----Original Message----- From: DRAGE, Keith (Keith)=20
> >>>>>> [mailto:drage@alcatel-lucent.com] Sent: 13. kes=E4kuuta 2007
> >>>>>> 22:25 To: Jonathan Rosenberg; IETF SIP List Subject: RE:
> >>>>>> [Sip] UA Loose Routing
> >>>>>>=20
> >>>>>> (As WG chair)
> >>>>>>=20
> >>>>>> As well as agreeing to charter the item, we actually agreed
> >>>>>>=20
> >>>>> at IETF#67
> >>>>>> that this draft was a suitable candidate to fulfil that
> >>>>> charter item.
> >>>>>> Therefore unless issues are raised with this, I would
> >>> request the
> >>>>>> editor (as Jonathan is editor) to submit the next
> >>> version of this
> >>>>>> document as draft-ietf-sip-ua-loose-route-00.
> >>>>>>=20
> >>>>>> Regards
> >>>>>>=20
> >>>>>> Keith
> >>>>>>=20
> >>>>>>> -----Original Message----- From: Jonathan Rosenberg=20
> >>>>>>> [mailto:jdrosen@cisco.com] Sent: Wednesday, June 13, 2007
> >>>>>>> 4:02 AM To: IETF SIP List Subject: [Sip] UA Loose=20
> Routing Issue=20
> >>>>>>> #1: Backwards
> >>>> compatibility
> >>>>>>> I've recently revised my draft on UA loose routing. You can
> >>>>>> pick it up
> >>>>>>> here:
> >>>>>>>=20
> >> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loo
> >>>>>>> se-route-01.txt
> >>>>>>>=20
> >>>>>>> This draft is meant to fulfill one of our charter items:=20
> >>>>>>> Feb 2008	  	Delivering request-URI and=20
> parameters to UAS
> >>>>>>> via proxy to IESG (PS)
> >>>>>>>=20
> >>>>>>> As folks may recall, the GRUU spec used to have these
> >>>>> things called
> >>>>>>> grids, which allowed a UA to add its own parameters to the
> >>>>>> GRUU before
> >>>>>>> handing it out. Several GRUU revisions back, this function
> >>>>>> got yanked
> >>>>>>> because it was recognized that the problem was more general
> >>>>>> than GRUU,
> >>>>>>> and would be useful for AOR as well.
> >>>>>>>=20
> >>>>>>> Two IETFs back I presented this draft briefly during a
> >>>>> SIP session.
> >>>>>>> There was a lot of interest and people liked it
> >>>> architecturally,
> >>>>>>> however there were two open issues that needed to get
> >>>>>> resolved. This
> >>>>>>> mail is meant to start discussion on one of them -
> >> backwards
> >>>>>>> compatibility.
> >>>>>>>=20
> >>>>>>> THe mechanism basically changes the way proxies handle
> >>>> requests.
> >>>>>>> Instead of rewriting the request URI for out-of-dialog
> >>>> requests,
> >>>>>>> they'll push a route header and leave the r-uri alone.
> >>>>>>> This
> >>>>>> allows the
> >>>>>>> original R-URI and any parameters it contains to get
> >>>>>> delivered to the
> >>>>>>> UAS.
> >>>>>>>=20
> >>>>>>> So, the main issue is, what kinds of backwards
> >>>>>> compatibility problems
> >>>>>>> will this cause. The draft provides a basic mechanism
> >>>>> whereby a UA
> >>>>>>> includes an option tag in the REGISTER, and the registrar
> >>>>>>>=20
> >>>>> indicates
> >>>>>>> that it is using this mechanism or not. So we can
> >>>> negotiate this
> >>>>>>> between a UA and registrar easily. Now, the problem is with=20
> >>>>>>> intermediate proxies. If there is an edge proxy between the
> >>>>>> UA and its
> >>>>>>> registrar, it will now see requests targeted to the UA,
> >>>>>> which contain
> >>>>>>> a Route header beyond itself, even though it thinks its the
> >>>>>> edge proxy
> >>>>>>> and the last point of contact before the UA. Technically,
> >>>>>>>=20
> >>>>>> if this edge
> >>>>>>> proxy is compliant to rfc3261 it would work. However, the
> >>>>>>>=20
> >>>>>> question is
> >>>>>>> whether proxies in the wild would properly work in
> >> this case.
> >>>>>>> If folks can comment based on their own products, or their=20
> >>>>>>> experiences, on whether they think this would work, or
> >>>>>> concerns they
> >>>>>>> have, that would be great. THere are solutions that would
> >>>>>>>=20
> >>>>>> allow us to
> >>>>>>> disable the mechanism when intermediate proxies don't
> >>>>>> support it, but
> >>>>>>> they are more complex and I'd rather avoid it.
> >>>>>>>=20
> >>>>>>> THanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D.
> >>>>>>> 600
> >>> Lanidex Plaza
> >>>>>>> Cisco Fellow
> >> Parsippany, NJ
> >>>>>>> 07054-2711 Cisco Systems jdrosen@cisco.com
> >>>>>>> FAX:
> >>>>> (973) 952-5050
> >>>>>>> http://www.jdrosen.net                         PHONE:
> >>>>> (973) 952-5000
> >>>>>>> http://www.cisco.com
> >>>>>>>=20
> >>>>>>>=20
> >>>>>>> _______________________________________________ Sip=20
> mailing list =20
> >>>>>>> https://www1.ietf.org/mailman/listinfo/sip
> >>>>>>> This list is for NEW development of the core SIP
> >> Protocol Use
> >>>>>>> sip-implementors@cs.columbia.edu for questions on
> >>>> current sip Use
> >>>>>>> sipping@ietf.org for new developments on the
> >>> application of sip
> >>>>>>=20
> >>>>>> _______________________________________________ Sip=20
> mailing list =20
> >>>>>> https://www1.ietf.org/mailman/listinfo/sip This list=20
> is for NEW=20
> >>>>>> development of the core SIP Protocol Use=20
> >>>>>> sip-implementors@cs.columbia.edu for questions on
> >>> current sip Use
> >>>>>> sipping@ietf.org for new developments on the
> >> application of sip
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20


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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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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		: A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg, et al.
	Filename	: draft-ietf-sip-consent-framework-02.txt
	Pages		: 29
	Date		: 2007-7-5
	
The Session Initiation Protocol (SIP) supports communications across
   many media types, including real-time audio, video, text, instant
   messaging, and presence.  In its current form, it allows session
   invitations, instant messages, and other requests to be delivered
   from one party to another without requiring explicit consent of the
   recipient.  Without such consent, it is possible for SIP to be used
   for malicious purposes, including amplification, and DoS (Denial of
   Service) attacks.  This document identifies a framework for consent-
   based communications in SIP.

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Maybe this is na=EFve, but doesn't this alter the operator trust model?=20

I always envisioned P-A-ID (as an example) as a way for a home network =
to tell a 3rd party service "trust this one, they are authenticated", as =
long as the home network and the service had a relationship. Having an =
edge proxy in a visited domain telling the home network "trust this one" =
seems to turn things around.

Steve.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Tuesday, July 03, 2007 8:40 PM
To: Steve Dotson
Cc: Dean Willis; IETF SIP List; DRAGE,Keith (Keith)
Subject: Re: [Sip] Certificate authentication in SIP



Steve Dotson wrote:
> Hi Jonathan,
>=20
> The use case we are looking at is where the registrar is separate from =

> the edge proxy.

Sure. I think that is a valid case.

> While the edge proxy could validate the UA has access to a cert issued =

> by a trusted root, it doesn't necessarily mean the UA has a valid=20
> subscription to network services.

No. But, such a proxy could assert the identity it verified.

> The edge proxy may/would not
> have access to this type of registration data, thus the need to=20
> authenticate from UA to registrar.

You lost me here. With mutual TLS, the client can authenticate itself to =
the edge proxy. The edge proxy can assert the authenticated identity =
towards the registrar, ala P-A-ID or even better, SIP Identity. THe =
registrar then authorizes access, and rejects the request if the user is =
not authorized.

-Jonathan R.

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com



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inline.

Dean Willis wrote:
> 
> On Jul 3, 2007, at 9:38 PM, Jonathan Rosenberg wrote:
> 
>> inline.
>>
>> Dean Willis wrote:
>>
>>>> So what? I think thats what ought to happen. I'd like to see some 
>>>> specific use cases where this can't work with the edge proxy 
>>>> performing the authentication. Keep in mind, we are talking about 
>>>> *certificate* authentication; that doesn't (by definition) required 
>>>> any kind of pre-arranged secret - only a common root CA.
>>> My edge proxy might be provided by MCI, but I might be using a chat 
>>> service from another provider overseas. This other provider might be 
>>> willing to trust the certificate for authentication (given that I 
>>> have a signed cert from MCI, who in turn has a signed CA cert from a 
>>> recognized op-level CA). However, the chat provider might not be 
>>> willing to just accept P-Asserted-Identity from MCI as an 
>>> authentication mechanism, since said provider does not have a trusted 
>>> peering arrangement with MCI and therefore ANYBODY could spoof my 
>>> P-Asserted-Identity.
>>
>> In a case where your chat provider has no relationship whatsoever with 
>> this edge proxy, for what purpose would you be connecting to it in the 
>> first place? You should connect directly to your chat provider 
>> offering the service. Then, if you have a cert - great - authenticate 
>> to it using mutual TLS.
>>
> 
> Well, I might be doing "outbound" with my edge proxy in order to get 
> some help from it for relaying messaging back through my NAT.

Why can't you do that with the edge proxy in your home providers network?

> Or it 
> might be a QoS-controlling proxy ala PacketCable. Or a 
> firewall-control-proxy . . .

Ah, well QoS control is much better handled through policy server 
peering than SIP connectivity. And it works for non-SIP things. Why is 
it that SIP is the vehicle for getting QoS in a visited network? Is this 
problem not more generic?

> 
> The chat provider might not just be on the other side of my edge proxy, 
> but on the other side of my home serving proxy too. And the certificate 
> auth stuff would appear to be able to traverse that proxy as well.

I don't follow the picture you have in mind.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Steve Dotson wrote:
> Maybe this is naïve, but doesn't this alter the operator trust model?
> 
> 
> I always envisioned P-A-ID (as an example) as a way for a home
> network to tell a 3rd party service "trust this one, they are
> authenticated", as long as the home network and the service had a
> relationship. Having an edge proxy in a visited domain telling the
> home network "trust this one" seems to turn things around.

P-A-ID asserts identity between providers with trust, yes that is true. 
SIP Identity works with lesser degrees of trust, but if you don't trust 
the signing domain, of course it doesn't work.

However, my point is, one of the two following scenarios always apply:

1. the edge proxy is trusted by the home proxy, in which case you can 
use mutual TLS to authenticate the client to the edge proxy, and then 
P-A-ID or SIP Identity from edge proxy to home proxy, *OR*

2. there is no trust at all between visited and home networks. In this 
case, you shouldn't be connectign to a visited proxy in the first place. 
COnnect to the home proxy in your home network. Then, case 1 applies again.


In other words, the entire motivation behind this draft is based on an 
assumption that a client needs to connect to its actual proxy through an 
intermediate proxy that no one trusts. SO many things get broken in that 
model. So, don't do it! This idea of a visited proxy with whom I have no 
relationship, is totally broken. ALl of the reasons people list for 
wanting them, we have far better solutions for (NAT traversal, firewall 
traversal, QoS). I'll note further that QoS can never work in a model 
without trust.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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inline.

Christer Holmberg (JO/LMF) wrote:
> Hi,
> 
>> My concern with P-Called-Party-ID is similar to my concern on 
>> History-Info. On History-Info, the draft states:
>> 
>> Firstly, it would cause the Request-URI to be relegated to nothing 
>> more than an indicator of the next hop for the request, identical 
>> exactly to the purpose of the Route header field. This results in 
>> two things in the SIP specification which do exactly the same
>> thing. Worse still, this is not just for some small feature of SIP
>> (where such duplication might not be a big deal), but rather, it
>> would be a duplication of SIP's primary function - routing of a
>> call towards a target.
>> 
>> This concern applies exactly to P-Called-Party-ID. Indeed, its even
>> worse, since now we would have, in a request:
>> 
>> Request-URI To P-Called-Party-ID History-Info Route
>> 
>> and I challenge you to explain to implementors what the differences
>> are. What would happen when they are inconsistent?
>> 
>> My objective was to reduce this count. Indeed I was hopeful that
>> UA-loose-route would allow us to deprecate P-Called-Party-ID, not
>> the other way around. WIth UA-loose-route, we end up with:
>> 
>> Request-URI: identifies the target Route: tells you how to get to
>> the target To: tells you the original target History info: sequence
>> of identities of targets
> 
> If people can't explain to their implementors the meaning of
> different SIP headers maybe THAT is something we should concentrate
> on, and if needed write a draft about.

Its not a question of documentation. Its that we would in fact have 
multiple headers which mean the same thing.

> 
>> which is much cleaner.
> 
> If we would re-do SIP I am sure lots of things could be done cleaner.
> 
> 
> P-Called-Party-ID does not change the current routing mechanism, and
> it does not have the potential issue with proxies (and possible also
> SBCs) that you have mentioned. And the proxy re-writing the R-URI can
> use the mechanism no matter if the UA supports it or not.

Sure, but its useless unless the UA supports it. So my point is, in all 
cases, we are talking about changes to UA and to proxy to do something 
useful here. The UA-loose-route mechanism doesn't require any changes 
beyond those two.

I do agree, that UA-loose-route requires a discovery mechanism, so the 
proxy knows whether to do this or not. But that is not difficult.

> 
> (And, naturally the mechanism would of course be backward compatible
> with entities and networks that already support P-Called-Party-ID.)
> 
>> So, functionally, P-Called-Party-ID could work. Any number of
>> things *can* work.
> 
> Yes, but P-Called-Party-ID is a standardized mechanism, made for this
> purpose, which most likely does work.

We have lots of P-headers that already exist. That doesn't mean we 
shouldn't bother to provide an architecturally better solution.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Francois

in-line

At 11:29 AM 6/18/2007, Francois Audet wrote:
>Hi James,
>
>It might be useful to have some wording on the limitations of using 
>a response to carry a Resource-Priority header in a response.

Do you have any text (or direction) to offer that you would want to 
see in this ID?


>Specifically, a UAS can not use a response to carry the 
>Resource-Priority in cases where the Resource-Priority "absolutely 
>has" to be delivered to the  UAC. It is therefore only useable for 
>cases where a forking proxy  may discard the response (i.e., forward 
>another instead) without breaking anything.

I have to admit, I do not understand this use-case (as it is 
written), since I'm not sure what you are referring to when you say 
"..."absolutely has" to be delivered to the  UAC".  Right now, RPH 
isn't in any response, so when did it become a situation of 
"..."absolutely has" to be delivered...".

Remind me what HEFP is again...

>In other words,you need to think about HEFP before using this draft.
>
>In some scenarios, this is probably the desired behavior.
>
>In other scenarios, it may not acceptable. So instead, for those scenarios,
>the terminating side would have to send a new request with a 
>Resource-Priority
>instead of using a response.
>
>
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Saturday, June 16, 2007 14:14
> > To: sip@ietf.org
> > Subject: [Sip] New draft modifying RFC 4412 for Responses
> >
> > SIP WG
> >
> > Here is a new ID I've written because of implementation
> > considerations to allow a SIP Resource-Priority header (RPH)
> > in responses, which RFC 4412 currently disallows, assuming
> > only that stateful devices will use RPH.  There have been
> > many requests to have this "only stateful devices"
> > restriction relaxed, in certain scenarios.  This ID does
> > this, and proposes how IANA is changed accordingly wrt RPH.
> > This ID does nothing else.
> >
> > Comments are appreciated.
> >
> > >A New Internet-Draft is available from the on-line Internet-Drafts
> > >directories.
> > >
> > >         Title           : Allowing SIP Resource Priority Header in
> > > SIP Responses
> > >         Author(s)       : J. Polk
> > >         Filename        : draft-polk-sip-rph-in-responses-00.txt
> > >         Pages           : 6
> > >         Date            : 2007-6-15
> > >
> > >    The Session Initiation Protocol (SIP) Resource Priority Header
> > >    (RPH), in its current form, is ignored in SIP responses.
> >  This was a
> > >    design choice during RFC 4412's development. This is now
> > considered
> > >    a bad design choice in certain scenarios.  This document corrects
> > >    RFC 4412's communications model by optionally allowing a
> > SIP server
> > >    or user agent client to process the Resource-Priority Header in a
> > >    response.
> > >
> > >A URL for this Internet-Draft is:
> > >http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-res
> > ponses-00.
> > >txt
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


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Henning

Regarding Conveyance -08 and the new Geolocation-Error header that 
you want to replace the Warning header code(s) function (section 3.4 
of the -07 ID)...

What fields do you want in the header?

#1 error-code,
#2 text-string explaining what error it is
#?

Do you need anything else?

Warning had/has the node ID of the sending entity. Do you want that 
in the new Geolocation-Error header?  Do you really see a purpose in 
having that piece there?

Attempting to put the final pieces of -08 together, and this new 
header is less than straightforward in writing it.

Is this viable ABNF for what you want?

Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
                         *(COMMA locationErrorValue))
locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
numeric-code       =  decimal-string
geoloc-error-param =  "code" EQUAL error-text-string
                        / generic-param ; (from RFC 3261)
error-text-string  =  string

(anyone) let me know what you think

Thanks
James


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Subject: RE: [Sip] Certificate authentication in SIP
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(1) Will the the P-A-ID syntax require modification to carry UA's =
certificate related information (Obtained during mutual TLS) from edge =
proxy to home proxy or registrar ?
(2) Edge proxy utilizing TLS offloaders.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Thursday, July 05, 2007 3:04 PM
To: Steve Dotson
Cc: IETF SIP List; DRAGE,Keith (Keith); Dean Willis
Subject: Re: [Sip] Certificate authentication in SIP



Steve Dotson wrote:
> Maybe this is na=EFve, but doesn't this alter the operator trust =
model?
>=20
>=20
> I always envisioned P-A-ID (as an example) as a way for a home network =

> to tell a 3rd party service "trust this one, they are authenticated",=20
> as long as the home network and the service had a relationship. Having =

> an edge proxy in a visited domain telling the home network "trust this =

> one" seems to turn things around.

P-A-ID asserts identity between providers with trust, yes that is true.=20
SIP Identity works with lesser degrees of trust, but if you don't trust =
the signing domain, of course it doesn't work.

However, my point is, one of the two following scenarios always apply:

1. the edge proxy is trusted by the home proxy, in which case you can =
use mutual TLS to authenticate the client to the edge proxy, and then =
P-A-ID or SIP Identity from edge proxy to home proxy, *OR*

2. there is no trust at all between visited and home networks. In this =
case, you shouldn't be connectign to a visited proxy in the first place. =

COnnect to the home proxy in your home network. Then, case 1 applies =
again.


In other words, the entire motivation behind this draft is based on an=20
assumption that a client needs to connect to its actual proxy through an =

intermediate proxy that no one trusts. SO many things get broken in that =

model. So, don't do it! This idea of a visited proxy with whom I have no =

relationship, is totally broken. ALl of the reasons people list for=20
wanting them, we have far better solutions for (NAT traversal, firewall=20
traversal, QoS). I'll note further that QoS can never work in a model=20
without trust.

-Jonathan R.



--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Hi Fredrik,

1) In this version I added text in the current Security Consideration
to help clarify this.  See the diffs in Section 13 here:

http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-outbound.diff.html

I thought I already sent you a message about this topic, but in any
case I will post my response here.  The goal is to make sure that *no
new attacks* are made possible by the introduction of this extension.
An active attacker can hijack your TCP connection before you add
outbound.  If you want to prevent active attacks, you need TLS,
whether you are using outbound or not.  The most important thing to
mention about the security of outbound is that is at least as secure
as SIP without outbound, and that the security gets quite good when
you add Digest.

2) When we originally talked about Algorithm 1 vs. Algorithm 2, there
was some attack we discussed that caused us to say that hijacking is
possible without the additional security in Algorithm 2.  I spoke with
Cullen about this today, and neither or us could recall the specific
attack.  It is possible that Algorithm 2 only prevents attacks that
are irrelevant based on the principle I mention in point 1 above, but
I want to make sure that we go back and look at the specific cases
that were mentioned on the list.

thanks,
-rohan


On 7/5/07, Fredrik Thulin <ft@it.su.se> wrote:
> DRAGE, Keith (Keith) wrote:
> > (As WG chair)
> >
> > We now have a new version of outbound. There hasn't exactly been a
> > flurry of comments since it was posted - does that mean it is now ready
> > for WGLC.
> >
> > It has agenda time at the next IETF meeting, with a view to starting
> > WGLC not long after that meeting. If you believe there are issues with
> > the current version of this document, then you need to post them to the
> > list, so that they can be either resolved there, or addressed in the
> > meeting.
>
> I never received an explanation of what kind of attacks could be avoided
> by an edge proxy using an algorithm that generates flow tokens that
> "cannot be modified by attackers" (section 5.2 of outbound-09).
>
> This was the last e-mail in the thread the last time I brought it up on
> the list :
>
> http://www1.ietf.org/mail-archive/web/sip/current/msg18336.html.
>
> This leads me to think that either
>
>    1) the security threats haven't been addressed properly
>
> or
>
>    2) that there in fact was no security threat so the requirement for an
>       edge proxy to generate flows that "cannot be modified by attackers"
>       is just unnecessary, and should therefor be removed
> or
>
>    3) there really _is_ something that gets significantly better by this
>       requirement, and it should simply be explained in the draft.
>
> /Fredrik
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From sip-bounces@ietf.org Thu Jul 05 21:32:20 2007
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James,

I think your #1 and #2 covers it, in addition to an identifier that  
points to the location element that suffers from the problem  
identified. (The latter was discussed, if I recall correctly, but  
seems to be missing from your syntax proposal.)

I don't feel strongly about identifying the originator of the error,  
but it might be useful to have an extensible parameter list, in  
standard SIP header fashion, so that we can easily add more debugging  
information later.

Henning

On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:

> Henning
>
> Regarding Conveyance -08 and the new Geolocation-Error header that  
> you want to replace the Warning header code(s) function (section  
> 3.4 of the -07 ID)...
>
> What fields do you want in the header?
>
> #1 error-code,
> #2 text-string explaining what error it is
> #?
>
> Do you need anything else?
>
> Warning had/has the node ID of the sending entity. Do you want that  
> in the new Geolocation-Error header?  Do you really see a purpose  
> in having that piece there?
>
> Attempting to put the final pieces of -08 together, and this new  
> header is less than straightforward in writing it.
>
> Is this viable ABNF for what you want?
>
> Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>                         *(COMMA locationErrorValue))
> locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
> numeric-code       =  decimal-string
> geoloc-error-param =  "code" EQUAL error-text-string
>                        / generic-param ; (from RFC 3261)
> error-text-string  =  string
>
> (anyone) let me know what you think
>
> Thanks
> James



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Rai, Anupam (Anupam) wrote:
> 
> (1) Will the the P-A-ID syntax require modification to carry UA's
> certificate related information (Obtained during mutual TLS) from
> edge proxy to home proxy or registrar ? 

Yes. You'd get nothing but the identity. Do you need more?


> (2) Edge proxy utilizing TLS
> offloaders.

The offloader would need to pass the identity to the server. If it acted 
as a simple SIP proxy in addition to doing TLS offload, it would work 
fine. There are other non-standard ways of passing this identity also.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: [Sip] Re: Conveyance -08 and Geolocation-Error header
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At 08:31 PM 7/5/2007, Henning Schulzrinne wrote:
>James,
>
>I think your #1 and #2 covers it, in addition to an identifier that
>points to the location element that suffers from the problem
>identified. (The latter was discussed, if I recall correctly, but
>seems to be missing from your syntax proposal.)

this is the Warning header's nodeID part. I wanted to know if you 
wanted that to remain in this new header. I sounds like you do.

Question - what's the difference between "...he location element that 
suffers from the problem identified..." mentioned above and how you 
"... don't feel strongly about identifying the originator of the 
error". Aren't they the same node?

If not, I need to understand, as I define what ID goes into this header.

Another point - should this error header only be in the 424 (Bad 
Location Information) response, or all 4XX responses, allowing a 
transaction to be successful - but without the location piece of the 
information exchange being good?

This was discussed, but I don't think the discussion ever concluded.


>I don't feel strongly about identifying the originator of the error,
>but it might be useful to have an extensible parameter list, in
>standard SIP header fashion, so that we can easily add more debugging
>information later.

I agree with this last point. Do you want something more than the 
existing "generic-param (from RFC 3261)" I have there now?


>Henning
>
>On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:
>
>>Henning
>>
>>Regarding Conveyance -08 and the new Geolocation-Error header that
>>you want to replace the Warning header code(s) function (section
>>3.4 of the -07 ID)...
>>
>>What fields do you want in the header?
>>
>>#1 error-code,
>>#2 text-string explaining what error it is
>>#?
>>
>>Do you need anything else?
>>
>>Warning had/has the node ID of the sending entity. Do you want that
>>in the new Geolocation-Error header?  Do you really see a purpose
>>in having that piece there?
>>
>>Attempting to put the final pieces of -08 together, and this new
>>header is less than straightforward in writing it.
>>
>>Is this viable ABNF for what you want?
>>
>>Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>>                         *(COMMA locationErrorValue))
>>locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
>>numeric-code       =  decimal-string
>>geoloc-error-param =  "code" EQUAL error-text-string
>>                        / generic-param ; (from RFC 3261)
>>error-text-string  =  string
>>
>>(anyone) let me know what you think
>>
>>Thanks
>>James


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From sip-bounces@ietf.org Thu Jul 05 21:53:53 2007
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Let me try to explain:

Let's say proxy P1 inserts location information L1, and user agent U1  
inserts L2. I want to identity L1 and L2, not necessarily the IP  
addresses or host names for proxy P1 or user agent U1 (which may or  
may not be unique, if NATs are involved).

The reason is simple - some element may insert location information,  
and the poor sender will try to figure out an error for location  
information it has never inserted.

I think this should be in all responses, as calls may succeed even if  
the location information, or one piece of it, happens to be bad. I  
still want to know that the call may have been routed without taking  
the location information into consideration, say, and why.

The 3261 generic-param is fine with me.

Henning

On Jul 5, 2007, at 9:45 PM, James M. Polk wrote:

> Question - what's the difference between "...he location element  
> that suffers from the problem identified..." mentioned above and  
> how you "... don't feel strongly about identifying the originator  
> of the error". Aren't they the same node?
>
> If not, I need to understand, as I define what ID goes into this  
> header.
>
> Another point - should this error header only be in the 424 (Bad  
> Location Information) response, or all 4XX responses, allowing a  
> transaction to be successful - but without the location piece of  
> the information exchange being good?
>
> This was discussed, but I don't think the discussion ever concluded.
>



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At 08:53 PM 7/5/2007, Henning Schulzrinne wrote:
>Let me try to explain:
>
>Let's say proxy P1 inserts location information L1, and user agent U1
>inserts L2. I want to identity L1 and L2, not necessarily the IP
>addresses or host names for proxy P1 or user agent U1 (which may or
>may not be unique, if NATs are involved).

<light goes on>

this is important... many of us had realized this was a problem, I 
just didn't realize this new header solved it (yet).  It will now!


>The reason is simple - some element may insert location information,
>and the poor sender will try to figure out an error for location
>information it has never inserted.
>
>I think this should be in all responses,

all?  even 5XXs and 6XXs?  I'm have a pragmatic attack thinking how 
in the world a UAS can process enough of a request to figure out 
location is bad, but still error the message because of something it 
can't do more fundamentally with the request outside of 
location.  There aren't that many 5XX and 6XX responses, so I'm not 
fighting it - just trying to understand what you want.

I'm open to all 4XX (cuz I can see this working in most 4XX responses 
(like a double error of "this *and* that" were wrong with this 
request), which is why I offered it (instead of limiting this to just the 424).


>as calls may succeed even if
>the location information, or one piece of it, happens to be bad. I
>still want to know that the call may have been routed without taking
>the location information into consideration, say, and why.
>
>The 3261 generic-param is fine with me.

good!

Thanks!


>Henning
>
>On Jul 5, 2007, at 9:45 PM, James M. Polk wrote:
>
>>Question - what's the difference between "...he location element
>>that suffers from the problem identified..." mentioned above and
>>how you "... don't feel strongly about identifying the originator
>>of the error". Aren't they the same node?
>>
>>If not, I need to understand, as I define what ID goes into this
>>header.
>>
>>Another point - should this error header only be in the 424 (Bad
>>Location Information) response, or all 4XX responses, allowing a
>>transaction to be successful - but without the location piece of
>>the information exchange being good?
>>
>>This was discussed, but I don't think the discussion ever concluded.


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--NextPart

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

                                                                                           
        Title           : Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)
        Author(s)       : J. Rosenberg
        Filename        : draft-ietf-sip-acr-code-05.txt
        Pages           : 8
        Date            : 2007-07-05
                                                                                           
The Session Initiation Protocol (SIP) allows for users to make
anonymous calls.  However, users receiving such calls have the right
to reject them because they are anonymous.  SIP has no way to
indicate to the caller that the reason for call rejection was that
the call was anonymous.  Such an indication is useful to allow the
call to be retried without anonymity.  This specification defines a
new SIP response code for this purpose.
                                                                                           
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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           : A Hitchhiker's Guide to the Session Initiation Protocol (SIP)
        Author(s)       : J. Rosenberg
        Filename        : draft-ietf-sip-hitchhikers-guide-03.txt
        Pages           : 32
        Date            : 2007-07-05
                                                                                           
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 the specifications under
the SIP umbrella, briefly summarizes each, and groups them into
categories.
                                                                                           
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt
                                                                                           
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I've just submitted a new version of the hitchhikers guide. You can pick 
it up at:
http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt

The changes from -02 are:


* connected-identity -> RFC4916
* added record-route-fix and grouped it with fork-loop-fix into a new
item under SIP core called "essential corrections"
* added subnot-etags under event packages
* added sip-sips to core specs

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


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>
> all?  even 5XXs and 6XXs?  I'm have a pragmatic attack thinking how  
> in the world a UAS can process enough of a request to figure out  
> location is bad, but still error the message because of something  
> it can't do more fundamentally with the request outside of  
> location.  There aren't that many 5XX and 6XX responses, so I'm not  
> fighting it - just trying to understand what you want.
>

I'm not a friend of being overly prescriptive and trying to predict  
all cases, including future status codes. In most cases, inserting  
the information may not make sense, but it is always harmless, even  
if it goes along with a 5xx or 6xx response. After all, both types of  
errors may occur at the same time. If the location information is  
broken, I want to be told about it, regardless of what other problems  
the SIP request had. (Plus, the location error information may have  
been inserted along the way by a proxy trying to use it for location  
routing. The UAS getting the location error indication shouldn't just  
swallow this information - it's useful to the UAC.)

Henning






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I've submitted an update to the acr-code draft. You can pick it up at:
http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-05.txt

This draft addresses AD comments raised during AD evaluation. The 
primary change is that, previously, the specification indicated that a 
P-A-ID header field with an anonymous URI in it would count as an 
anonymous request. However, P-A-ID doesn't reflect what the user wants, 
which is strictly in the From. Consequently, PAID is really irrelevant 
here. So, this condition was removed.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: RE: [Sip] Certificate authentication in SIP
Date: Thu, 5 Jul 2007 20:27:21 -0600
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=20
inline

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Thursday, July 05, 2007 6:33 PM
> To: Rai, Anupam (Anupam)
> Cc: Steve Dotson; IETF SIP List; DRAGE,Keith (Keith); Dean Willis
> Subject: Re: [Sip] Certificate authentication in SIP
>=20
>=20
>=20
> Rai, Anupam (Anupam) wrote:
> >=20
> > (1) Will the the P-A-ID syntax require modification to carry UA's=20
> > certificate related information (Obtained during mutual=20
> TLS) from edge=20
> > proxy to home proxy or registrar ?
>=20
> Yes. You'd get nothing but the identity. Do you need more?
>=20
[Anupam Rai]
Identity as asserted by certificate presented by UA during mutual TLS or
identity as established by edge proxy after consulting some
authentication service/database ?
>=20
> > (2) Edge proxy utilizing TLS
> > offloaders.
>=20
> The offloader would need to pass the identity to the server.=20
> If it acted as a simple SIP proxy in addition to doing TLS=20
> offload, it would work fine. There are other non-standard=20
> ways of passing this identity also.
>=20
> -Jonathan R.
>=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20


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inline:

Rai, Anupam (Anupam) wrote:
>  
> inline
> 
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
>> Sent: Thursday, July 05, 2007 6:33 PM
>> To: Rai, Anupam (Anupam)
>> Cc: Steve Dotson; IETF SIP List; DRAGE,Keith (Keith); Dean Willis
>> Subject: Re: [Sip] Certificate authentication in SIP
>>
>>
>>
>> Rai, Anupam (Anupam) wrote:
>>> (1) Will the the P-A-ID syntax require modification to carry UA's 
>>> certificate related information (Obtained during mutual 
>> TLS) from edge 
>>> proxy to home proxy or registrar ?
>> Yes. You'd get nothing but the identity. Do you need more?
>>
> [Anupam Rai]
> Identity as asserted by certificate presented by UA during mutual TLS or
> identity as established by edge proxy after consulting some
> authentication service/database ?

As asserted by the certificate. In other words, the edge proxy would 
validate the client cert, extract the subjectAltName which presumably 
has something like joe@example.com somewhere and then shoves that into a 
P-Asserted-ID. Or if it matches the From it would use RFC 4474 and sign 
the request.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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At 09:14 PM 7/5/2007, Henning Schulzrinne wrote:

>>all?  even 5XXs and 6XXs?  I'm have a pragmatic attack thinking how
>>in the world a UAS can process enough of a request to figure out
>>location is bad, but still error the message because of something
>>it can't do more fundamentally with the request outside of
>>location.  There aren't that many 5XX and 6XX responses, so I'm not
>>fighting it - just trying to understand what you want.
>
>I'm not a friend of being overly prescriptive and trying to predict
>all cases, including future status codes. In most cases, inserting
>the information may not make sense, but it is always harmless, even
>if it goes along with a 5xx or 6xx response. After all, both types of
>errors may occur at the same time. If the location information is
>broken, I want to be told about it, regardless of what other problems
>the SIP request had. (Plus, the location error information may have
>been inserted along the way by a proxy trying to use it for location
>routing. The UAS getting the location error indication shouldn't just
>swallow this information - it's useful to the UAC.)

fair -- the text is changed to allow it in all responses

New topic -- Do we want at all to try and tackle mapping of which 
location goes with which error, to make sure each receiver of the 
response pays attention or ignores what they are supposed to (wrt 
locationValues)?

If this is a good idea, would a header parameter sequence number 
(mapped between Geolocation header and Geolocation-Error header) do 
the trick? I think we'll just need to make sure the numbers don't 
replicate another sequence number within the same Call-ID, and not be 
universally unique through space and time.

This might be a little too much to take on this late (to the 
deadline) or it just might be too silly of an idea *or* do you think 
the host ID within the Geolocation-Error header might be enough to 
cover this idea.


>Henning


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Subject: RE: [Sip] Certificate authentication in SIP
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inline=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Thursday, July 05, 2007 7:38 PM
> To: Rai, Anupam (Anupam)
> Cc: Steve Dotson; IETF SIP List; DRAGE,Keith (Keith); Dean Willis
> Subject: Re: [Sip] Certificate authentication in SIP
>=20
> inline:
>=20
> Rai, Anupam (Anupam) wrote:
> > =20
> > inline
> >=20
> >> -----Original Message-----
> >> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> >> Sent: Thursday, July 05, 2007 6:33 PM
> >> To: Rai, Anupam (Anupam)
> >> Cc: Steve Dotson; IETF SIP List; DRAGE,Keith (Keith); Dean Willis
> >> Subject: Re: [Sip] Certificate authentication in SIP
> >>
> >>
> >>
> >> Rai, Anupam (Anupam) wrote:
> >>> (1) Will the the P-A-ID syntax require modification to carry UA's=20
> >>> certificate related information (Obtained during mutual
> >> TLS) from edge
> >>> proxy to home proxy or registrar ?
> >> Yes. You'd get nothing but the identity. Do you need more?
> >>
> > [Anupam Rai]
> > Identity as asserted by certificate presented by UA during=20
> mutual TLS=20
> > or identity as established by edge proxy after consulting some=20
> > authentication service/database ?
>=20
> As asserted by the certificate. In other words, the edge=20
> proxy would validate the client cert, extract the=20
> subjectAltName which presumably has something like=20
> joe@example.com somewhere and then shoves that into a=20
> P-Asserted-ID. Or if it matches the From it would use RFC=20
> 4474 and sign the request.

[Anupam Rai]
It would be nice to include some more information related to certificate
presented by UA. At a minimum identity of the UA's certificate issuer is
required.

>=20
> -Jonathan R.
>=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20


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>
> New topic -- Do we want at all to try and tackle mapping of which  
> location goes with which error, to make sure each receiver of the  
> response pays attention or ignores what they are supposed to (wrt  
> locationValues)?
>
> If this is a good idea, would a header parameter sequence number  
> (mapped between Geolocation header and Geolocation-Error header) do  
> the trick? I think we'll just need to make sure the numbers don't  
> replicate another sequence number within the same Call-ID, and not  
> be universally unique through space and time.
>

Agreed.

> This might be a little too much to take on this late (to the  
> deadline) or it just might be too silly of an idea *or* do you  
> think the host ID within the Geolocation-Error header might be  
> enough to cover this idea.

I'm not sure what format the hostID has. If IP address, you could end  
up with two 192.168.0.1, from different hosts, which would be rather  
confusing.


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From sip-bounces@ietf.org Thu Jul 05 23:23:44 2007
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The IESG has received a request from the Session Initiation Protocol WG 
(sip) to consider the following document:

- 'Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)
'
   <draft-ietf-sip-acr-code-05.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2007-07-19. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-05.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=14158&rfc_flag=0



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At 10:06 PM 7/5/2007, Henning Schulzrinne wrote:

>>New topic -- Do we want at all to try and tackle mapping of which
>>location goes with which error, to make sure each receiver of the
>>response pays attention or ignores what they are supposed to (wrt
>>locationValues)?
>>
>>If this is a good idea, would a header parameter sequence number
>>(mapped between Geolocation header and Geolocation-Error header) do
>>the trick? I think we'll just need to make sure the numbers don't
>>replicate another sequence number within the same Call-ID, and not
>>be universally unique through space and time.
>
>Agreed.
>
>>This might be a little too much to take on this late (to the
>>deadline) or it just might be too silly of an idea *or* do you
>>think the host ID within the Geolocation-Error header might be
>>enough to cover this idea.
>
>I'm not sure what format the hostID has. If IP address, you could end
>up with two 192.168.0.1, from different hosts, which would be rather
>confusing.

Conveyance -07 gave a Warning hostID of alice.example.com as the format.

I agree this shouldn't likely ever be an IP address - due to your 
exact example problem.



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Hi;
After a brief reading I found this document very useful as a guide to 
the SIP maze.

Just 2 typos regarding 3325 and 3323 (pages 5 and 9). They define the 
header P-Asserted-Identity and not P-Asserted-ID.

Thank you for putting together this document

Diego B

Jonathan Rosenberg wrote:
> I've just submitted a new version of the hitchhikers guide. You can 
> pick it up at:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt 
>
>
> The changes from -02 are:
>
>
> * connected-identity -> RFC4916
> * added record-route-fix and grouped it with fork-loop-fix into a new
> item under SIP core called "essential corrections"
> * added subnot-etags under event packages
> * added sip-sips to core specs
>
> Thanks,
> Jonathan R.


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Hi Jonathan,

A comment on the emergency services. The text does not address any of 
the citizen-to-authority emergency work.
Hence, the term "emergency services and the mentioning of 911 might be a 
bit misleading

You might want to change the section heading to something like 
"Preemption and Precedence".

Ciao
Hannes




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From sip-bounces@ietf.org Fri Jul 06 06:09:21 2007
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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-consent-framework-02.txt 
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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(As WG chair)=20

I am in the process of preparing the publication request on this
document to IESG, and this revision is to address various issues (none
of them particularly technical) identified during that process.

If you identify any issues with this document a comment within the next
couple of days will possibly receive attention (before the document
deadline) else it may have to wait for the IESG last call to be handled.

Regards

Keith

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Thursday, July 05, 2007 10:15 PM
> To: i-d-announce@ietf.org
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-consent-framework-02.txt=20
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> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> This draft is a work item of the Session Initiation Protocol=20
> Working Group of the IETF.
>=20
> 	Title		: A Framework for Consent-Based=20
> Communications in the Session Initiation Protocol (SIP)
> 	Author(s)	: J. Rosenberg, et al.
> 	Filename	: draft-ietf-sip-consent-framework-02.txt
> 	Pages		: 29
> 	Date		: 2007-7-5
> =09
> The Session Initiation Protocol (SIP) supports communications across
>    many media types, including real-time audio, video, text, instant
>    messaging, and presence.  In its current form, it allows session
>    invitations, instant messages, and other requests to be delivered
>    from one party to another without requiring explicit consent of the
>    recipient.  Without such consent, it is possible for SIP to be used
>    for malicious purposes, including amplification, and DoS (Denial of
>    Service) attacks.  This document identifies a framework=20
> for consent-
>    based communications in SIP.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-consent-fra
> mework-02.txt
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> Internet-Drafts are also available by anonymous FTP. Login=20
> with the username "anonymous" and a password of your e-mail=20
> address. After logging in, type "cd internet-drafts" and then=20
> "get draft-ietf-sip-consent-framework-02.txt".
>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html or=20
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-consent-framework-02.txt".
> =09
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant=20
> mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
>=20
> Below is the data which will enable a MIME compliant mail=20
> reader implementation to automatically retrieve the ASCII=20
> version of the Internet-Draft.
>=20


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From sip-bounces@ietf.org Fri Jul 06 11:41:35 2007
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Technical:

- Related to Open Issue 3, there is also the question of what the
Content-Disposition of a response body is.  For SDP answers, "session"
makes sense, but the body may contain control information that doesn't
fall under "session" or "render" cleanly.

- Section 4.1:  "If the handling of a 'multipart/alternative' body is
required, the UA MUST set the 'handling' parameter of the
'multipart/alternative' body and to the last body part within the
'multipart/alternative' to 'required'."

Is there a typo here?  The whole point of multipart/alternative is
that one need only understand one component, and tagging the last
component 'required' contradicts this.

It would seem that even if the whole body was "required", any
individual component would be "optional".  (Though of course, the
reader is obliged to understand at least one component.)  Indeed, the
handling of any component of a multipart/alternative is implied by its
membership in the multipart/alternative and need not be stated.

- In regard to error reporting, perhaps we should use the Warning header
and describe the problem in natural language, rather than trying to
provide errors that can be processed automatically.

- In regard to other multipart types:

   This specification recommends support for 'multipart/mixed' and
   'multipart/alternative'.  At present, there are no SIP extensions
   that use different 'multipart' subtypes such as parallel [RFC2046],
   digest [RFC2046], or related [RFC2387].  If such extensions were to
   be defined in the future, their authors would need to make sure
   (e.g., by using an option-tag or by other means) that entities
   receiving those 'multipart' subtypes were able to process them.  As
   stated earlier, UAs treat unknown 'multipart' subtypes as 'multipart/
   mixed'.

Since there is a standard fallback, an extension does not need to
provide an option-tag as long as it is willing to have its multipart
type processed as multipart/mixed.  So it would be more accurate to
say "If such extensions were to be defined in the future, their
authors might need to provide UAs a way (e.g., by using an option-tag)
to make sure that entities receiving those 'multipart' subtypes were
able to process them as such."

Editorial:

- There is a tendency of the terminology to overlook the fact that all
SIP bodies are MIME bodies, and to use "MIME body" as synonymous with
"MIME multipart body".  E.g., in section 1, "Multipart message bodies
are encoded using the MIME (Multipurpose Internet Mail Extensions)
[RFC2045] format." is literally correct but suggests that
non-multipart message bodies somehow aren't encoded using MIME.  In
reality, Content-Type is a MIME header, etc.

Dale


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Hmm. I've been behind on reading. I was having a side conversation with 
James on this, and it looks like some of that would be worth sharing 
here. James was asking me for ideas of how to handle this "identifier".

The following is what I last sent to James:

Paul Kyzivat wrote:
> What do you expect will happen as a result of identifying the source of the error?
> 
> Do you expect that the source will itself recognize that it has made a mistake and do something about it? Or do you expect some other element to keep notes on which nodes do a bad job of supplying addresses?
> 
> Also, do you expect the node that inserts the address to remember what address it inserted when it gets the error response? Or some other node other than the one that inserted it to remember it?
> 
> The answer to these questions impacts the kind of solution that is appropriate.
> 
> If the UAS is the one to detect the problem, and also the one to log the problem for future use, then it must be possible to actually determine the address of the source of the address. An index into the Via list might be adequate for that. Or, the "inserted-by" could be changed to actually include the address of the node doing the insertion.
> 
> If the node doing the insertion is responsible for tracking its own failures, then its probably bad to assume that it will remember what it inserted. So the actual offending address should probably be included in the error response. In that case, there is probably something in that address that will allow the offender to recognize that it was the source of the problem.
> 
> If some other node (other than the UAS and the offending node) is responsible for logging the problem, then it will certainly want the offending address as well. So again it should be included in the response. And again, there probably isn't need of anything else.
> 
> At least that's my take based on what little info I have.

	Paul

Henning Schulzrinne wrote:
> James,
> 
> I think your #1 and #2 covers it, in addition to an identifier that 
> points to the location element that suffers from the problem identified. 
> (The latter was discussed, if I recall correctly, but seems to be 
> missing from your syntax proposal.)
> 
> I don't feel strongly about identifying the originator of the error, but 
> it might be useful to have an extensible parameter list, in standard SIP 
> header fashion, so that we can easily add more debugging information later.
> 
> Henning
> 
> On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:
> 
>> Henning
>>
>> Regarding Conveyance -08 and the new Geolocation-Error header that you 
>> want to replace the Warning header code(s) function (section 3.4 of 
>> the -07 ID)...
>>
>> What fields do you want in the header?
>>
>> #1 error-code,
>> #2 text-string explaining what error it is
>> #?
>>
>> Do you need anything else?
>>
>> Warning had/has the node ID of the sending entity. Do you want that in 
>> the new Geolocation-Error header?  Do you really see a purpose in 
>> having that piece there?
>>
>> Attempting to put the final pieces of -08 together, and this new 
>> header is less than straightforward in writing it.
>>
>> Is this viable ABNF for what you want?
>>
>> Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>>                         *(COMMA locationErrorValue))
>> locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
>> numeric-code       =  decimal-string
>> geoloc-error-param =  "code" EQUAL error-text-string
>>                        / generic-param ; (from RFC 3261)
>> error-text-string  =  string
>>
>> (anyone) let me know what you think
>>
>> Thanks
>> James
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Fixed. Thanks!

-Jonathan R.

Diego B wrote:
> Hi;
> After a brief reading I found this document very useful as a guide to 
> the SIP maze.
> 
> Just 2 typos regarding 3325 and 3323 (pages 5 and 9). They define the 
> header P-Asserted-Identity and not P-Asserted-ID.
> 
> Thank you for putting together this document
> 
> Diego B
> 
> Jonathan Rosenberg wrote:
>> I've just submitted a new version of the hitchhikers guide. You can 
>> pick it up at:
>> http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt 
>>
>>
>> The changes from -02 are:
>>
>>
>> * connected-identity -> RFC4916
>> * added record-route-fix and grouped it with fork-loop-fix into a new
>> item under SIP core called "essential corrections"
>> * added subnot-etags under event packages
>> * added sip-sips to core specs
>>
>> Thanks,
>> Jonathan R.
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Dale.Worley@comcast.net wrote:

> - Section 4.1:  "If the handling of a 'multipart/alternative' body is
> required, the UA MUST set the 'handling' parameter of the
> 'multipart/alternative' body and to the last body part within the
> 'multipart/alternative' to 'required'."

The general recommendation for processing multipart/alternative is to 
process the *last* part that you are capable of understanding. Thus the 
least capable recipient will be processing the first and ignoring the rest.

> Is there a typo here?  The whole point of multipart/alternative is
> that one need only understand one component, and tagging the last
> component 'required' contradicts this.
> 
> It would seem that even if the whole body was "required", any
> individual component would be "optional".  (Though of course, the
> reader is obliged to understand at least one component.)  Indeed, the
> handling of any component of a multipart/alternative is implied by its
> membership in the multipart/alternative and need not be stated.

I think clearly the goal is that if handling of the 
multipart/alternative itself is "required", then *one* of the 
alternatives is to be understood and processed, but no particular one is 
required. If handling for the multipart is "optional" then either one of 
the parts may be processed, or none.

The question is: what is the way to express that?

I think I'm with Dale, that the handling parameter on the 
multipart/alternative specifies the intent, that all the contained parts 
should be optional, and the semantics of processing the alternative 
implies that one of the parts will be.

	Paul


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Marcos Dytz wrote:
 >
 > I agree on the limitations described below by Jeff. My concern was with
 > the emphasis on authentication throughout the draft, but I had in mind
 > that it was actually targeting that particular point (in the same way
 > that sip-paying targets payment).

Ok.


 > Nevertheless, are there ideas to develop new drafts on further cases for
 > SIP/SAML?

Yes. Hannes has a few "in the queue".

 > In my personal opinion, the protocol offers many technical

which protocol?

 > advantages (and flexibilities) that could be required for more general
 > use cases.



 > By adopting and developing new cases, the overall protocol
 > would only be gaining a larger market share and could very well adopt
 > some synergy among the different areas.

here, by "..the overall protocol..", you mean "SIP+SAML"?


 > As for Sebastian, I have the impression that he would like to see more
 > integration and deployment details. I had problems to fulfill the gaps
 > on integration while working on my thesis (e.g., enveloping SAML
 > assertions in SIP message - usually workflows target the transaction
 > state and avoid mentioning how to converge that with SIP - I did my
 > implementation in the way I thought best, but every developer is free to
 > choose different formats if specification does not define these little
 > details).

But you weren't implementing what is specified in draft-ietf-sip-saml-02 were you?

Do you have pointers to any documentation or papers about your work?  Or the 
code? :)


 > As for your final points, agree to limitations due to complexity and
 > partial fulfilment of RFC4484. If further work is to be applied on the
 > draft, I believe the best action would be to leave the draft as it is
 > and start another work from scratch.

Ok, fine by me.

thanks again for your feedback,

=JeffH




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Subject: [Sip] ICE tutorial in Chicago [DO NOT REPLY]
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(Apologies to multiple recipients)
(DO NOT REPLY)

WHAT:     Tutorial on Interactive Connectivity Establishment (ICE)
       (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-16.txt)
           [-17 should be available shortly]

WHEN:     Tuesday, July 24, 2007, 1145-1245
           (lunch logistics are in progress and will be announced ASAP)

WHERE:    Red Lacquar

WHO:      Anyone that would like to learn more about ICE.

WHY:      ICE, developed by the mmusic working group, has been
           defined as the IETF tool for NAT traversal for SIP-based
           media. It is also seeing increased usage for traversal
           for other kinds of application traffic - anything worrying
           about any-to-any communications. In this tutorial, we will
           overview the background of the problem, consider the
           requirements driving ICE, and provide an overview of
           how ICE works. An understanding of SIP is not required.
           The tutorial will be based on the pending -17 version.

RSVP:     Please send a note to me with the Subject line "ice-is-nice"
           (mailto:jdrosen@cisco.com?Subject=ice-is-nice) so I can get a
           count of the number of participants for lunch purposes.


!DO NOT REPLY TO THIS NOTE!
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: [Sip] wrt SIP Saml Review by Jiri Kuthan - "missing scenarios"
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Jiri Kuthan <jiri@iptel.org> wrote:
 > I'm personally missing scenarios which
 >
 > - use redirection as opposed to proxy mode. IMO, it is beneficial to have
 >   the AS operated in 3xx mode for better scalability.

Perhaps you could explain a bit more what you mean here because it isn't all 
that clear what the differences would be. In poking around thru, say, RFC3665 
section "3.6.  Session via Redirect and Proxy Servers with SDP in ACK", it 
isn't clear to me that there is necessarily a salient difference between it's 
scenario and the one depicted in draft-ietf-sip-saml-02 fig 1.

Here's RFC3665 Session via Redirect and Proxy Servers..

    Alice        Redirect Server     Proxy 3             Bob
      |                |                |                |
      |   INVITE F1    |                |                |
      |--------------->|                |                |
      |     302 F2     |                |                |
      |<---------------|                |                |
      |     ACK F3     |                |                |
      |--------------->|                |                |
      |     INVITE F4                   |                |
      |-------------------------------->|    INVITE F5   |
      |             100  F6             |--------------->|
      |<--------------------------------|      180 F7    |
      |             180 F8              |<---------------|
      |<--------------------------------|                |
      |                                 |     200 F9     |
      |             200 F10             |<---------------|
      |<--------------------------------|                |
      |             ACK F11             |                |
      |-------------------------------->|     ACK F12    |
      |                                 |--------------->|
      |                Both Way RTP Media                |
      |<================================================>|
      |                                 |     BYE F13    |
      |             BYE F14             |<---------------|
      |<--------------------------------|                |
      |             200 F15             |                |
      |-------------------------------->|     200 F16    |
      |                                 |--------------->|
      |                                 |                |


..and here's -sip-saml-02 fig 1..


      +--------+           +--------------+          +--------+
      |Alice@  |           |Authentication|          | Bob@   |
      |example |           |Service       |          |example2|
      |.com    |           |@example.com  |          |.com    |
      +---+----+           +------+-------+          +---+----+
          |                       |                      |
          |      INVITE           |                      |
          |---------------------->|                      |
          | From:alice@foo.com    |                      |
          |                       |                      |
          |  407 Proxy auth. req. |                      |
          |<----------------------|                      |
          |     Challenge         |                      |
          |                       |                      |
          |       ACK             |                      |
          |---------------------->|                      |
          |                       |                      |
          | INVITE w/authn creds  |                      |
          |---------------------->|                      |
          |                       | INVITE               |
          |                       | w/Identity header    |
          |                       |--------------------->|
          |                       | and Identity-Info    |
          |                       |                      |
          |                       | HTTP GET SAML assn   |
          |                       |<==================== |
          |                       | and domain cert      |
          |                       |                      |
          |                       | HTTP 200 OK + assn   |
          |                       |=====================>|
          |                       | and domain cert      |
          |            200 OK     |                      |
          |<----------------------+----------------------|
          |                       |                      |


..so it seems that given the way in which RFC4474 works, either of the Redirect 
server or Proxy 3 in the former figure could instigate authn of the user/UAC, 
whereupon things would proceed more-or-less as depicted in the latter.

yes/no?


 > - use SAML-by-value, in addition to SAML-by-reference. I personally find
 >   the SAML-by-reference quite non-real-time and harder-to-scale too.

Yes, this is not-yet-well-resolved piece of the architecture. Given present 
restrictions wrt SIP intermediaries only being able to add SIP headers to a SIP 
request message -- it offhand would have to be the UAC that embeds a SAML 
assertion into a SIP request it is originating.

We (SIP WG) should perhaps think about this and whether we think it is a 
practical use case that would indeed see some use. Below is one example of such 
an approach (there's various ways to approach its details). Is this an approach 
we should pursue in addition to the one presently spec'd in -sip-saml-2 ?


 > Perhaps there are arguments why proxy+by/reference is the best thing, but
 > if that's the case I think they should be mentioned in the document.

Yes, ok. Part of the rationale was that this sort of approach will work with 
present-day unmodified UACs, and all of the workable 
"include-SAML-assertion-by-value" approaches that we could think of involved 
UAC enhancements to support the new funtionality (this of course skirts the 
controversial sip-proxy-placing-BLOB-values-in-SIP-header approach, e.g. using 
a data: URL).

So we felt that specifying the draft narrowly, and focusing on enhancements to 
proxy-tier functionality was the way to go for now.

thoughts?  HTH,

=JeffH

-------
Fig 4: "caller-driven, authn+attr, SAML Assertion embedded in INVITE"

        Caller explicitly executes <samlp:AuthnRequest> protocol, i.e.
          "caller-driven" in that caller stipulates at least some of info
          that is to be embodied in resulting assertion.
        Caller places SAML Assertion in INVITE msg body; Assertion has an
          <saml:AuthnStatement> and optionally a <saml:AttributeStatement>. 

        Callee obtains SAML Assertion directly from INVITE message.

        protocol message count (only include msgs w/CSeq=N+1):

          Caller-to-AS:  3
          AS-to-callee:  1

        Protocols employed: SIP and SAML (via HTTP/SOAP)

        NOTE:    The interaction between caller and the AS is
                 in the style of the "SAML Token Service", which is
                 specified in draft form in section 

                 "3.3 SAML Token Service Profile" of [9].

                 Caller and AS must speak at least one flavor of SAML protocol.

                 Relying party needs only to be capable of understanding
                 SAML assertion. Does not need to employ any SAML protocols.


[9] SAML 2.0 Single Sign-On with Constrained Delegation
     (and SAML Token Service Proposal (Shib design note))
http://shibboleth.internet2.edu/docs/draft-cantor-saml-sso-delegation-01.pdf



                                                     Callee's UA or
                             Caller's Domain's      Callee's Domain's
         Caller               AS & SIP Proxy           SIP Proxy
        +--------+           +--------------+          +--------+
        |Alice@  |           |Authentication|          | Bob@   |
        |example |           |Service       |          |example2|
        |.com    |           |@example.com  |          |com     |
        |        |           |    (AS)      |          |        |
        +---+----+           +------+-------+          +---+----+
-   -      |                       |                      |
^   ^      |      INVITE           |                      |
|   |      |---------------------->|                      |
|          | From:alice@foo.com    |                      |
|   C      | To:sip:bob@example.com|                      |
|   S      |                       |                      |
|   e      |  407 Proxy auth. req. |                      |
|   q      |<----------------------|                      |
|          |     Challenge         |                      |
|   =      |                       |                      |
|          |                       |                      |
|   N      |                       |                      |
|          |                       |                      |
|   |      |      ACK              |                      |
|   V      |---------------------->|                      |
|   -      |                       |                      |
|          |                       |                      |
|          | [Alice marshalls      |                      |
|          |  authn credentials    |                      |
|          |  in resp to challenge]|                      |
|          |                       |                      |
|          |                       |                      |
|    -     |Request Authn+Attr Assn|                      |
|    ^     |via Authn Request Prot |                      |
|    .     |======================>|                      |
|    .     | <samlp:AuthnRequest>  |                      |
|    .     |  <samlpext:RespondTo> |                      |
|    .     |     bob@example2.com  | [AS is configured    |
|    .     |                       |  OOB to "know" which |
|    .     | Assn returned         |  "profile attributes"|
|    .     |<======================|  to return in Assns. |
|    .     | <samlp:Response>      |  Relying party meta- |
|    .     |  <saml:Assertion>     |  data could be also  |
|    .     |   <saml:Subject>      |  employed. Or SAML   |
|    .     |    <saml:NameID>      |  request protocol    |
|    .     |      Alice@example.com|  extension could be  |
|    .     |   <saml:AuthnStatement>  used.]              |
|    .     |   <saml:AttrStatement>|                      |
|    .     |      foo=bar          |                      |
|    .     |                       |                      |
      .     |                       |                      |
D    .     |                       |                      |
      .     | INVITE + authorization|                      |
I    |     | header w/marshalled creds                    |
      |     |---------------------->|                      |
A    |     | From:alice@foo.com    |                      |
      |     | To:sip:bob@example.com|                      |
L    |     |  <saml:Assertion>     |                      |
      |     |   <saml:Subject>      |                      |
O    |     |    <saml:NameID>      |                      |
      |     |      Alice@example.com|                      |
G    |     |   <saml:AuthnStatement>                      |
      |     |   <saml:AttrStatement>|                      |
|    |     |      foo=bar          |                      |
|    |     |                       |                      |
|    |     |                       |                      |
|    |     |                       |                      |
|          |                       |INVITE + authorization|
|    C     |                       |header w/marshalled creds
|    S     |                       |--------------------->|
|    e     |                       | From:alice@foo.com   | [this would
|    q     |                       | To:sip:bob@example.com  thus convey an
|          |                       |  <saml:Assertion>    |  "unsolicited
|    =     |                       |   <saml:Subject>     |     response" as
            |                       |    <saml:NameID>     |   discussed in
            |                       |      Alice@example.com   [5] & [3]. ]
            |                       |   <saml:AuthnStatement>
|          |                       |   <saml:AttrStatement>
|    N     |                       |     foo=bar          |
|          |                       |                      |
|    +     |                       |                      |
|          |                       |                      |
|    1     |                       |                      |
|          |                       |                      |
|    |     |                       |                      |
|    |     |            200 OK     |                      |
.    V     |<----------------------+----------------------|
.    -     |                       |                      |
.
.
V




-------
end






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From sip-bounces@ietf.org Fri Jul 06 20:27:04 2007
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Richard Barnes wrote:
 >
 > 1. It's never made clear what entity is the Subject of the conveyed
 > assertions.  This could be a critical distinction, e.g., if one party
 > assumes that it's the UAC being discussed, and the other the user of the
 > UAC.

section "6.1.4.1.2.  Element: <Subject>" stipulates..

    The <Subject> element SHOULD contain both a <NameID> element and a
    <SubjectConfirmation> element.

    The value of the <NameID> element MUST be the same as the Address of
    Record (AoR) value used in computing the "signed-identity-digest"
    which forms the value of the Identity header.  See Section 9 of
    [RFC4474].


So, whatever is (currently) being stuffed into AoRs also goes into the 
Subject/NameID element.

Does that answer this?


 > 2. Section 9 needs a discussion the nature of the assurances provided by
 > SAML assertions.

Yes, though more specifically, the nature of the assurances provided by a SAML 
assertion constructed, and referenced by a SIP request, as specified in this spec.


 > In general, in order for it to be verifiable that the
 > SAML assertion was issued by the Authentication Service, there must be
 > proof that the holder of the private key used to create the
 > Identity-Info header.

Agreed, though this is in the spec.


 >There are several ways to do this:
 > 2.a. The SAML assertion is signed with the appropriate private key

This is specified in section "6.1.4.  Assertion Profile Description/Overall 
SAML assertion profile requirements:"...

       The SAML assertion MUST be signed by the same key as used to sign
       the contents of the Identity header field.  Signing of SAML
       assertions is defined in Section 5.4 of [OASIS.saml-core-2.0-os].


 > 2.c. The Identity-Info is integrity-protected from the point where it is
 > added by the Authentication Service until it reaches the UAS.

This is a basic feature of rfc4474.


 > 3. The list of threats and countermeasures in Section 9 is incomplete.
 > For instance:
 > 3.a. There's no discussion of threats of attacks by elements of the SIP
 > signaling path (e.g., manipulating the URI in the Identity-Info header)
 > 3.b. The anti-replay mechanism should reference the expiry elements
 > built into the assertion (e.g., the NotOnOrAfter attribute)

yes, ok.


 > 4. The document isn't clear on whether or not the assertions MUST be
 > signed by the issuer.

This is specified in section "6.1.4.  Assertion Profile Description/Overall 
SAML assertion profile requirements:"...

       The SAML assertion MUST be signed by the same key as used to sign
       the contents of the Identity header field.  Signing of SAML
       assertions is defined in Section 5.4 of [OASIS.saml-core-2.0-os].

..tho perhaps we should clarify that the wielder of this key is "the issuer".


 >                       Section 6.1.5, step 3 seems to imply that the
 > assertion will be signed, but Section 9.3 seems to assume that this
 > isn't necessarily the case.

hm, I don't see that. RFC4474 goes into gory detail wrt the signing of the 
identity headers and I didn't want to replicate it here. Plus as noted above, 
the -sip-saml-02 says assn MUST be signed (with same key as signed Identity 
header field).


 > IMHO, it should be that the assertion MUST
 > be signed by the same key used to generate the identity digest.

agreed ;)


 > 5. No mention is made of the possible sensitivity of the contents of
 > SAML assertions.  It's possible that these assertions could contain
 > sensitive information about the Subject, such as statements about what
 > identities the subject holds.

ah, indeed. good catch.


 > There should be some discussion about
 > what measures should be taken in this case, e.g., the use of HTTPS and
 > authentication of dereferencing clients.

ok.



 > Section 6.1.5, Step 3:
 > Section 5.4 of the indicated reference doesn't say anything about how
 > signatures are verified.

well, that's not exactly true. There's info therein about certain first-order 
properties assns must have in order to verify correctly and such, so it still 
needs to be referenced imho. But yes, it doesn't provide a step-by-step guide 
for doing so.


 > Instead, this should reference Section 3.2 of
 > the XMLdsig document referenced.

will be more explicit about the references into XMLdsig -- presently it's just 
a blanket ref.


 > Also is this TODO resolved?

we're working on it. I think we should alloc a new sip error value, as has been 
proposed by Hannes. just need to choose one.



 > Section 6.1.5, Step 5:
 > Specify how this verification is done, e.g., by verifying that the
 > signed-identity-digest and the signature on the SAML assertion are both
 > verifiable with the key in the cert in the assertion.

agreed, good catch.


 > Section 6.1.5, Step 7:
 > The relevant field in the X.509 certificate are Subject and Subject
 > Alternative Name (not Issuer and Issuer Alternative Name).  The Issuer
 > of the assertion is the Subject of the certificate.

ah! yeah, another good catch. Also we should probably reference section 13.4 of 
rfc4474.

thanks for your review,

=JeffH





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 > -------- Original Message --------
 > Subject: [Sip] SIP Saml Review by Marcos Dytz
 > Date: Mon, 18 Jun 2007 10:39:20 +0200
 > From: Tschofenig, Hannes <hannes.tschofenig@nsn.com>
 > To: SIP IETF <sip@ietf.org>
 > CC: ext Marcos Dytz <marcos.dytz@gmail.com>
 >
 > 6) I believe that it is valid that together with signing part of the
 > message, some fields are hashed in order to improve security.

Hm, I'm not sure I understand. Can you please explain more fully?

thanks,

=JeffH




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From sip-bounces@ietf.org Fri Jul 06 20:54:32 2007
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 > -------- Original Message --------
 > Subject: 	[Sip] Comments draft-ietf-sip-saml-02
 > Date: 	Thu, 21 Jun 2007 16:47:46 +0200
 > From: 	Willekens, Marc <marc.willekens@nsn.com>
 > To: 	<sip@ietf.org>
 >
 >
 > [Page30] Security considerations
 >
 > Is it possible to redirect messages to an own authentication service by
 > manipulating the end-device?

hm, do you mean e.g. that the receiving proxy UAS might, acting as an AS, issue 
a "407 Proxy authentication request" back up the signaling path?


 > How can the called party trust the
 > authentication service assigned by the caller?

Well, this is more of an overall question and thus falls into the realm of "sip 
identity" rfc4474. I think that the best we have at this time is section "13.4. 
  Domain Names and Subordination" of rfc4474. Note that it references out to 
the Security Considerations section of rfc3280 "Internet X.509 Public Key 
Infrastructure Certificate and Certificate Revocation List (CRL) Profile".

Note also that rfc4474 sec 13.4 notes in it's 2nd-to-last paragraph that..

                                          "Some federations of service
    providers, for example, might only trust certificates that have been
    provided by a certificate authority operated by the federation."

perhaps that's the best we can do for now.

Perhaps we need to explain this better in -sip-saml-xx.

=JeffH




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My general take is that the source, on the return trip, should be  
able to identify location information it has inserted. Note that a  
single UA or proxy could theoretically insert two locations (e.g.,  
geo + civic) or reference and value, so some additional  
identification beyond the node may be necessary. If it weren't for  
various SBCs mucking with Via's, the Via solution sounds interesting,  
but maybe the simple host name is easiest.

The problem with the host name (or Via) is that the party discovering  
the error and the party inserting the location are generally *NOT*  
the same. The error is likely to be discovered far downstream, e.g.,  
by the UAS or the PSAP proxy. The UAS or PSAP proxy has no way to  
know what node inserted the location, since the Geolocation header  
field currently only says "endpoint" or "server".

Thus, another variation that allows host identification and avoids  
this problem:

<proposal>
Just define ;inserted-by to use the host name, and then use the same  
parameter in the error indication.
</proposal>

Henning

On Jul 6, 2007, at 2:27 PM, Paul Kyzivat wrote:

> Hmm. I've been behind on reading. I was having a side conversation  
> with James on this, and it looks like some of that would be worth  
> sharing here. James was asking me for ideas of how to handle this  
> "identifier".
>
> The following is what I last sent to James:
>
> Paul Kyzivat wrote:
>> What do you expect will happen as a result of identifying the  
>> source of the error?
>> Do you expect that the source will itself recognize that it has  
>> made a mistake and do something about it? Or do you expect some  
>> other element to keep notes on which nodes do a bad job of  
>> supplying addresses?
>> Also, do you expect the node that inserts the address to remember  
>> what address it inserted when it gets the error response? Or some  
>> other node other than the one that inserted it to remember it?
>> The answer to these questions impacts the kind of solution that is  
>> appropriate.
>> If the UAS is the one to detect the problem, and also the one to  
>> log the problem for future use, then it must be possible to  
>> actually determine the address of the source of the address. An  
>> index into the Via list might be adequate for that. Or, the  
>> "inserted-by" could be changed to actually include the address of  
>> the node doing the insertion.
>> If the node doing the insertion is responsible for tracking its  
>> own failures, then its probably bad to assume that it will  
>> remember what it inserted. So the actual offending address should  
>> probably be included in the error response. In that case, there is  
>> probably something in that address that will allow the offender to  
>> recognize that it was the source of the problem.
>> If some other node (other than the UAS and the offending node) is  
>> responsible for logging the problem, then it will certainly want  
>> the offending address as well. So again it should be included in  
>> the response. And again, there probably isn't need of anything else.
>> At least that's my take based on what little info I have.
>
> 	Paul
>
> Henning Schulzrinne wrote:
>> James,
>> I think your #1 and #2 covers it, in addition to an identifier  
>> that points to the location element that suffers from the problem  
>> identified. (The latter was discussed, if I recall correctly, but  
>> seems to be missing from your syntax proposal.)
>> I don't feel strongly about identifying the originator of the  
>> error, but it might be useful to have an extensible parameter  
>> list, in standard SIP header fashion, so that we can easily add  
>> more debugging information later.
>> Henning
>> On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:
>>> Henning
>>>
>>> Regarding Conveyance -08 and the new Geolocation-Error header  
>>> that you want to replace the Warning header code(s) function  
>>> (section 3.4 of the -07 ID)...
>>>
>>> What fields do you want in the header?
>>>
>>> #1 error-code,
>>> #2 text-string explaining what error it is
>>> #?
>>>
>>> Do you need anything else?
>>>
>>> Warning had/has the node ID of the sending entity. Do you want  
>>> that in the new Geolocation-Error header?  Do you really see a  
>>> purpose in having that piece there?
>>>
>>> Attempting to put the final pieces of -08 together, and this new  
>>> header is less than straightforward in writing it.
>>>
>>> Is this viable ABNF for what you want?
>>>
>>> Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>>>                         *(COMMA locationErrorValue))
>>> locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
>>> numeric-code       =  decimal-string
>>> geoloc-error-param =  "code" EQUAL error-text-string
>>>                        / generic-param ; (from RFC 3261)
>>> error-text-string  =  string
>>>
>>> (anyone) let me know what you think
>>>
>>> Thanks
>>> James
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip



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Date: Fri, 06 Jul 2007 23:24:58 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
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I have no skin in this game, so take my comments with a grain of salt...

Is it really reasonable to expect a proxy that inserts a location to 
remember what location it inserted, in enough detail to diagnose why it 
might have been messed up? I presume it would have to reconstruct what 
the location would have been based in information in the response. It 
may not be possible to reconstruct the exact same value, because the UAC 
may have moved in the meantime. I guess the inserted location would have 
to be by reference, but the URL of the reference may itself not be 
easily reconstructed, and multiple queries of it may not return the same 
result.

IMO it would be much clearer to include the location itself in the 
response. And if the location had been passed by reference, it might be 
better to pass it by value in the response, so that there is no 
possibility of the value changing.

OTOH that might result in divulging the location to nodes upstream of 
the one that included it. If it was passed by reference then the node 
that had inserted it could remove the reference in the response. But if 
it is passed back by value that isn't allowed.

The alternative is that the proxy will have to retain the location until 
the completion of the transaction. Maybe that's ok, but if its only to 
diagnose an occasional glitch I get it doesn't get done.

	Paul

Henning Schulzrinne wrote:
> My general take is that the source, on the return trip, should be able 
> to identify location information it has inserted. Note that a single UA 
> or proxy could theoretically insert two locations (e.g., geo + civic) or 
> reference and value, so some additional identification beyond the node 
> may be necessary. If it weren't for various SBCs mucking with Via's, the 
> Via solution sounds interesting, but maybe the simple host name is easiest.
> 
> The problem with the host name (or Via) is that the party discovering 
> the error and the party inserting the location are generally *NOT* the 
> same. The error is likely to be discovered far downstream, e.g., by the 
> UAS or the PSAP proxy. The UAS or PSAP proxy has no way to know what 
> node inserted the location, since the Geolocation header field currently 
> only says "endpoint" or "server".
> 
> Thus, another variation that allows host identification and avoids this 
> problem:
> 
> <proposal>
> Just define ;inserted-by to use the host name, and then use the same 
> parameter in the error indication.
> </proposal>
> 
> Henning
> 
> On Jul 6, 2007, at 2:27 PM, Paul Kyzivat wrote:
> 
>> Hmm. I've been behind on reading. I was having a side conversation 
>> with James on this, and it looks like some of that would be worth 
>> sharing here. James was asking me for ideas of how to handle this 
>> "identifier".
>>
>> The following is what I last sent to James:
>>
>> Paul Kyzivat wrote:
>>> What do you expect will happen as a result of identifying the source 
>>> of the error?
>>> Do you expect that the source will itself recognize that it has made 
>>> a mistake and do something about it? Or do you expect some other 
>>> element to keep notes on which nodes do a bad job of supplying 
>>> addresses?
>>> Also, do you expect the node that inserts the address to remember 
>>> what address it inserted when it gets the error response? Or some 
>>> other node other than the one that inserted it to remember it?
>>> The answer to these questions impacts the kind of solution that is 
>>> appropriate.
>>> If the UAS is the one to detect the problem, and also the one to log 
>>> the problem for future use, then it must be possible to actually 
>>> determine the address of the source of the address. An index into the 
>>> Via list might be adequate for that. Or, the "inserted-by" could be 
>>> changed to actually include the address of the node doing the insertion.
>>> If the node doing the insertion is responsible for tracking its own 
>>> failures, then its probably bad to assume that it will remember what 
>>> it inserted. So the actual offending address should probably be 
>>> included in the error response. In that case, there is probably 
>>> something in that address that will allow the offender to recognize 
>>> that it was the source of the problem.
>>> If some other node (other than the UAS and the offending node) is 
>>> responsible for logging the problem, then it will certainly want the 
>>> offending address as well. So again it should be included in the 
>>> response. And again, there probably isn't need of anything else.
>>> At least that's my take based on what little info I have.
>>
>>     Paul
>>
>> Henning Schulzrinne wrote:
>>> James,
>>> I think your #1 and #2 covers it, in addition to an identifier that 
>>> points to the location element that suffers from the problem 
>>> identified. (The latter was discussed, if I recall correctly, but 
>>> seems to be missing from your syntax proposal.)
>>> I don't feel strongly about identifying the originator of the error, 
>>> but it might be useful to have an extensible parameter list, in 
>>> standard SIP header fashion, so that we can easily add more debugging 
>>> information later.
>>> Henning
>>> On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:
>>>> Henning
>>>>
>>>> Regarding Conveyance -08 and the new Geolocation-Error header that 
>>>> you want to replace the Warning header code(s) function (section 3.4 
>>>> of the -07 ID)...
>>>>
>>>> What fields do you want in the header?
>>>>
>>>> #1 error-code,
>>>> #2 text-string explaining what error it is
>>>> #?
>>>>
>>>> Do you need anything else?
>>>>
>>>> Warning had/has the node ID of the sending entity. Do you want that 
>>>> in the new Geolocation-Error header?  Do you really see a purpose in 
>>>> having that piece there?
>>>>
>>>> Attempting to put the final pieces of -08 together, and this new 
>>>> header is less than straightforward in writing it.
>>>>
>>>> Is this viable ABNF for what you want?
>>>>
>>>> Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>>>>                         *(COMMA locationErrorValue))
>>>> locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
>>>> numeric-code       =  decimal-string
>>>> geoloc-error-param =  "code" EQUAL error-text-string
>>>>                        / generic-param ; (from RFC 3261)
>>>> error-text-string  =  string
>>>>
>>>> (anyone) let me know what you think
>>>>
>>>> Thanks
>>>> James
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
> 


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Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
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On Jul 6, 2007, at 11:24 PM, Paul Kyzivat wrote:

> Is it really reasonable to expect a proxy that inserts a location  
> to remember what location it inserted, in enough detail to diagnose  
> why it might have been messed up? I presume it would have to  
> reconstruct what the location would have been based in information  
> in the response. It may not be possible to reconstruct the exact  
> same value, because the UAC may have moved in the meantime. I guess  
> the inserted location would have to be by reference, but the URL of  
> the reference may itself not be easily reconstructed, and multiple  
> queries of it may not return the same result.
>
> IMO it would be much clearer to include the location itself in the  
> response. And if the location had been passed by reference, it  
> might be better to pass it by value in the response, so that there  
> is no possibility of the value changing.
>

I tend to agree that this would be better. The reference-to-value  
translation may not always be feasible: If a proxy inserts a location  
reference and a downstream proxy finds a problem, it can't insert a  
value (except by the data URL mechanism I have been mentioning...).


> OTOH that might result in divulging the location to nodes upstream  
> of the one that included it. If it was passed by reference then the  
> node that had inserted it could remove the reference in the  
> response. But if it is passed back by value that isn't allowed.
>
> The alternative is that the proxy will have to retain the location  
> until the completion of the transaction. Maybe that's ok, but if  
> its only to diagnose an occasional glitch I get it doesn't get done.

My assumption was that most locations will be inserted by the UAC  
rather than a proxy. In those cases, retaining the information is  
probably not a big deal, since the UAC will generally only have one.

In summary, I think returning as much information about the erroneous  
location information as possible seems best, i.e., both the location  
information and the party that inserted it. This avoids the "who the  
heck put that bogus location information in there?" problem.

In general, I think we have learned that we should help people with  
debugging distributed systems with distributed responsibility, as  
otherwise they'll either struggle or invent private mechanisms to  
include this information. (See SER for the the latter.)


>
> 	Paul
>



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Hi Erkki, et al.

I hope you are enjoying your summer holiday.  I'm assuming that Aki
and/or Miguel will join in the thread, per your other email.

I was hoping that Fran=E7ois Audet would reply to the time-keepalive
part of this thread, but I will reply based on one of the use cases he
described to me.  I will also talk a bit about keep-stun and
keep-crlf.

Comments inline.

On 6/28/07, Erkki.Koivusalo@nokia.com <Erkki.Koivusalo@nokia.com> wrote:
>
> Hi,
>
> I have today had a look into the fresh Outbound-09
> to see how it addresses the keepalive issues in comparison
> to the proposal I submitted originally in the beginning of
> March and resent it yesterday.
>
> My first questions are related to this statement:
>
>    Clients MUST support this CRLF keepalive.
>
> If so, what is the value of promoting the TCP Keepalive within
> SIP Outbound ? In which circumstancies an UA supporting TCP
> Keepalive would instead select sending CRLF keepalive ?

If the server did not add "timed-keepalives", and the UA is happy with
the value of the TCP keepalive timers, then the UA can use TCP
keepalive.

> Further on, I like this piece of text within Outbound-09:
>
>    The client needs to perform normal RFC 3263 [4] SIP DNS resolution on
>    the URI from the outbound-proxy-set to pick a transport.  Once a
>    transport is selected, the UA selects a keepalive approach that is
>    allowed for that transport ...
>
> That approach is elegant and consistent with RFC 3263.
>
> But I still dislike these pieces of text:
>
>    ... and that is allowed by the server based on the tags in the URI
>    from the outbound-proxy-set.
>
>    User Agents that form flows check if the configured URI they are
>    connecting to has a 'keep-crlf' URI parameter (defined in
>    Section 12).  If the parameter is present and the UA is not already
>    performing keepalives using another supported mechanism, the UA can
>    send keep alives as described in this section.
>
>    User Agents that form flows, check if the configured URI they are
>    connecting to has a 'keep-stun' URI parameter (defined in
>    Section 12).  If the parameter is present and the UA is not already
>    performing keepalives using another supported mechanism, the UA can
>    periodically perform keepalive checks by sending STUN [3] Binding
>    Requests over the flow as described in Section 8.
>
> Outbound-09 suggests that configured proxy URIs could contain
> 'keep-crlf' or 'keep-stun' parameters, with which the UAs could
> check whether the proxy supports the corresponding keepalive
> mechanism before starting to send such keepalives. Instead of
> all that I simply suggest using the 'ob' parameter for the proxy
> URI to indicate that the proxy supports SIP Outbound and all the
> keepalive mechanisms defined in SIP Outbound for those transports
> the proxy supports according to DNS.
>
> Rationale:
>
> 1. SIP Outbound anyway already mandates compliant proxies to
>    support all keepalive mechanisms (see section 5.4) so there is
>    no need to define separate flags for each of those mechanisms.
>    One single flag for Outbound support is sufficient indication.
>
> 2. The draft has already defined the 'ob' URI parameter and how the
>    proxy should use it. My proposal is to use the flag as defined
>    but additionally to bundle the semantics of separate keepalive
>    flags to this single 'ob' parameter.
>
> 3. The UA could check the presence of 'ob' parameter within the
>    proxy URI either
>         a) within its configured outbound-proxy-set; or
>         b) Path header received within 200 OK to REGISTER
>            (like suggested in section 8)
>
> 4. Simple is better: less flags to be introduced to IANA,
>    to the URIs or to the implementations. Resulting URIs
>    would also be shorter with this single and short parameter.

The 'ob' parameter is not appropriate if we continue to allow STUN
keepalives to be used without outbound.  Christer requested this and
the WG agreed to move the STUN keepalive definition to a separate
section of the draft.

We could certainly change 'keep-stun' and 'keep-crlf' to just 'keep'
(for example), but we had consensus that supporting keepalives and
supporting outbound should be kept orthogonal.  In addition, I think
using 'ob' in so many contexts would be confusing:
- in the first Path header it means that the first Edge Proxy added a flow =
token
- in the UACs Contact header it means that the UAC registered
successfully with outbound.  The first Edge Proxy can use this as a
hint to add a flow-token to its Record-Route.
Then you propose to add this one:
- in the outbound-proxy-set URIs it means that keepalives are supported.
- in the topmost Path header returned after registration it means that
keepalives are supported

Having 'keep-stun' and 'keep-crlf' is merely historic and I support
merging these into a single 'keep' parameter if we have consensus to
do so.


> Finally I do not find any value for the 'timed-keepalives'
> parameter. It was agreed in IETF68 but I was not attending
> the meeting and I have not seen the rationale for it anywhere.
> For the proxy point of view what is the difference whether or
> not the proxy URI contains this parameter ? If the parameter
> is within the URI, the proxy can be sure that it receives
> the keepalives with intervals defined in the spec but otherwise
> the UA could select longer intervals. However the spec does not
> specify any actions for the proxy to check how often it receives
> the keepalive messages so what is the point to have such a URI
> parameter ? Instead of that I would suggest dropping this
> parameter and either:

This parameter was a compromise between folks who want the client to
be able to send whenever it likes (typically less often, to conserve
battery power for example); and folks who want their server to be able
to track liveness of their clients.

The spec does not define what a proxy using timed-keepalives would do
with this information, but below are some examples.

1) The UA is used in a call center.  If a certain number of keepalives
are missed by a UA, the proxy assumes the UA could be down and only
forwards calls to that agent when no "live" clients are available.

2) If several keepalives are missed, the proxy declares the UA down
and deletes the flow.

3) Associated with the proxy is a management station that shows the
status of each UA connected to it.

I am not trying to argue for including these features.  I am just
providing an explanation of what folks might do with this parameter.

I would like to point out that this only works if the EP and the
registrar are coordinated at least via configuration.  A simple way to
do this in a cross-domain environment would be for the EPs to include
;keep in the outbound-proxy-set and for the registrar to add
;timed-keepalives to the Path header for the EP.

> a) mandate the UA always to use timers as defined in the draft
> or
> b) let the UA always to choose any timer values it wants to use
>    and make the timer values in the draft as recommendations

Choice a) certainly does not work well for battery operated devices,
and choice b) does not enable the use cases I described above.

> Option b) could be better if the UA later on has some
> mechanism (like STUN extension etc.) to ask the middleboxes
> to adjust their timer behaviour.

Discovering the topology between the UA and the EP is easy 90% of the
time and practically impossible 1% of the time.  The problem is that
the UA can't tell which situation it is in. Consequently, I won't hold
my breath for a deterministic solution to middlebox timer
configuration.

> Example of URIs for a proxy supporting all keepalive methods:
>
> Outbound-09:
> sip:pri.example.com;lr;keep-stun;keep-crlf;timed-keepalives;ob
>
> My proposal:
> sip:pri.example.com;lr;ob

Your first URI isn't quite correct.  If your proxy has no interest in
timed-keepalives, then it would omit the parameter.  The 'ob'
parameter does not appear in the outbound-proxy-set.  If we
consolidate keep-stun and keep-crlf, the situation looks pretty good.

outbound-proxy-set:
sip:pri.example.com;lr;keep-stun;keep-crlf  OR
sip:pri.example.com;lr;keep

Path:
sip:pri.example.com;lr;keep-stun;keep-crlf;ob
sip:pri.example.com;lr;keep;ob

I don't think this is quite as daunting.

thanks,
-rohan

> Just guess why I prefer the latter ;-) Any comments ?
>
> Regards,
>
> Erkki
>


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At 08:34 PM 7/6/2007, Henning Schulzrinne wrote:
>My general take is that the source, on the return trip, should be
>able to identify location information it has inserted. Note that a
>single UA or proxy could theoretically insert two locations (e.g.,
>geo + civic) or reference and value, so some additional
>identification beyond the node may be necessary. If it weren't for
>various SBCs mucking with Via's, the Via solution sounds interesting,
>but maybe the simple host name is easiest.
>
>The problem with the host name (or Via) is that the party discovering
>the error and the party inserting the location are generally *NOT*
>the same. The error is likely to be discovered far downstream, e.g.,
>by the UAS or the PSAP proxy. The UAS or PSAP proxy has no way to
>know what node inserted the location, since the Geolocation header
>field currently only says "endpoint" or "server".
>
>Thus, another variation that allows host identification and avoids
>this problem:
>
><proposal>
>Just define ;inserted-by to use the host name, and then use the same
>parameter in the error indication.
></proposal>

wouldn't this run into the same SBCs & B2BUAs mucking it up problem, 
thereby making this unreliable? I mean there could easily be 
domain-of-host scrubbing going on at borders, to include this field.

I'm not sure either if PSAPs reeeeally care if location was inserted 
by a UAC or proxy along the way - they just want something.  It 
sounded good when I first came up with the idea, but it is also 
merely informative in nature, and has zero integrity.

Also, if I go with your proposal, which seems easy to add, what 
strength do I put on the location inserter putting this in?  It seems 
like it ought to be something like  "...MUST insert inserter's 
host-id/name header parameter in the locationValue".

Matching this host-id/name into the GeolocationError header's 
locationErrorValue within the response seems easy then.

BTW Paul - there are lots of GeolocationError codes of various 
flavors, depending on what was wrong with the location when it was discovered.


>Henning
>
>On Jul 6, 2007, at 2:27 PM, Paul Kyzivat wrote:
>
>>Hmm. I've been behind on reading. I was having a side conversation
>>with James on this, and it looks like some of that would be worth
>>sharing here. James was asking me for ideas of how to handle this
>>"identifier".
>>
>>The following is what I last sent to James:
>>
>>Paul Kyzivat wrote:
>>>What do you expect will happen as a result of identifying the
>>>source of the error?
>>>Do you expect that the source will itself recognize that it has
>>>made a mistake and do something about it? Or do you expect some
>>>other element to keep notes on which nodes do a bad job of
>>>supplying addresses?
>>>Also, do you expect the node that inserts the address to remember
>>>what address it inserted when it gets the error response? Or some
>>>other node other than the one that inserted it to remember it?
>>>The answer to these questions impacts the kind of solution that is
>>>appropriate.
>>>If the UAS is the one to detect the problem, and also the one to
>>>log the problem for future use, then it must be possible to
>>>actually determine the address of the source of the address. An
>>>index into the Via list might be adequate for that. Or, the
>>>"inserted-by" could be changed to actually include the address of
>>>the node doing the insertion.
>>>If the node doing the insertion is responsible for tracking its
>>>own failures, then its probably bad to assume that it will
>>>remember what it inserted. So the actual offending address should
>>>probably be included in the error response. In that case, there is
>>>probably something in that address that will allow the offender to
>>>recognize that it was the source of the problem.
>>>If some other node (other than the UAS and the offending node) is
>>>responsible for logging the problem, then it will certainly want
>>>the offending address as well. So again it should be included in
>>>the response. And again, there probably isn't need of anything else.
>>>At least that's my take based on what little info I have.
>>
>>         Paul
>>
>>Henning Schulzrinne wrote:
>>>James,
>>>I think your #1 and #2 covers it, in addition to an identifier
>>>that points to the location element that suffers from the problem
>>>identified. (The latter was discussed, if I recall correctly, but
>>>seems to be missing from your syntax proposal.)
>>>I don't feel strongly about identifying the originator of the
>>>error, but it might be useful to have an extensible parameter
>>>list, in standard SIP header fashion, so that we can easily add
>>>more debugging information later.
>>>Henning
>>>On Jul 5, 2007, at 8:22 PM, James M. Polk wrote:
>>>>Henning
>>>>
>>>>Regarding Conveyance -08 and the new Geolocation-Error header
>>>>that you want to replace the Warning header code(s) function
>>>>(section 3.4 of the -07 ID)...
>>>>
>>>>What fields do you want in the header?
>>>>
>>>>#1 error-code,
>>>>#2 text-string explaining what error it is
>>>>#?
>>>>
>>>>Do you need anything else?
>>>>
>>>>Warning had/has the node ID of the sending entity. Do you want
>>>>that in the new Geolocation-Error header?  Do you really see a
>>>>purpose in having that piece there?
>>>>
>>>>Attempting to put the final pieces of -08 together, and this new
>>>>header is less than straightforward in writing it.
>>>>
>>>>Is this viable ABNF for what you want?
>>>>
>>>>Geolocation-Error  =  "Geolocation-Error" HCOLON (locationErrorValue
>>>>                         *(COMMA locationErrorValue))
>>>>locationErrorValue =  numeric-code *(SEMI geoloc-error-param)
>>>>numeric-code       =  decimal-string
>>>>geoloc-error-param =  "code" EQUAL error-text-string
>>>>                        / generic-param ; (from RFC 3261)
>>>>error-text-string  =  string
>>>>
>>>>(anyone) let me know what you think
>>>>
>>>>Thanks
>>>>James
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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>>>Use sip-implementors@cs.columbia.edu for questions on current sip
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>
> wouldn't this run into the same SBCs & B2BUAs mucking it up  
> problem, thereby making this unreliable? I mean there could easily  
> be domain-of-host scrubbing going on at borders, to include this  
> field.
>

Sure - any information could be removed. It just seems less likely.

> I'm not sure either if PSAPs reeeeally care if location was  
> inserted by a UAC or proxy along the way - they just want  
> something.  It sounded good when I first came up with the idea, but  
> it is also merely informative in nature, and has zero integrity.

I agree that the PSAP doesn't care who inserted the information and  
there's really not much difference between one inserted by a proxy  
and the end system, as far as the PSAP is concerned. The only one  
that cares is probably the caller (UAC), mostly if things go wrong.  
It can be really frustrating if some crucial information is inserted  
by some proxy you can't see, and you have no idea who did it and thus  
don't know if you should talk to your local provider or some other  
party closer to the destination. I don't know whether this is a  
realistic issue, given that most location information will be  
inserted either by the UAC or the outbound proxy.

Thus, maybe it would be sufficient to make the suitable extensibility  
hooks in both the Geolocation header and the error indication so that  
these can be added later, if we can't agree quickly on a suitable  
format.

>
> Also, if I go with your proposal, which seems easy to add, what  
> strength do I put on the location inserter putting this in?  It  
> seems like it ought to be something like  "...MUST insert  
> inserter's host-id/name header parameter in the locationValue".
>
> Matching this host-id/name into the GeolocationError header's  
> locationErrorValue within the response seems easy then.
>
> BTW Paul - there are lots of GeolocationError codes of various  
> flavors, depending on what was wrong with the location when it was  
> discovered.
>



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At 06:41 PM 7/7/2007, Henning Schulzrinne wrote:
>>I'm not sure either if PSAPs reeeeally care if location was
>>inserted by a UAC or proxy along the way - they just want
>>something.  It sounded good when I first came up with the idea, but
>>it is also merely informative in nature, and has zero integrity.
>
>I agree that the PSAP doesn't care who inserted the information and
>there's really not much difference between one inserted by a proxy
>and the end system, as far as the PSAP is concerned. The only one
>that cares is probably the caller (UAC), mostly if things go wrong.
>It can be really frustrating if some crucial information is inserted
>by some proxy you can't see, and you have no idea who did it and thus
>don't know if you should talk to your local provider or some other
>party closer to the destination. I don't know whether this is a
>realistic issue, given that most location information will be
>inserted either by the UAC or the outbound proxy.
>
>Thus, maybe it would be sufficient to make the suitable extensibility
>hooks in both the Geolocation header and the error indication so that
>these can be added later, if we can't agree quickly on a suitable
>format.

as much as anything, I'm thinking of going with your proposal to 
change "inserted-by=endpoint or server" to "inserted-by=host-id/name" 
so that transaction statefulness of location insertion is easier to 
recognize in a response.

This would blow up the idea of indicating if a locationValue was 
inserted by a endpoint or server... that was my point.

It also gets us away from tracking a sequence number (of what 
length?) and having to nail down all the parameters of that new value 
- consensus might be in 2009 on this one with a number



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From sip-bounces@ietf.org Mon Jul 09 02:32:29 2007
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Jonathan Rosenberg wrote:

>>>
>>> In a case where your chat provider has no relationship whatsoever 
>>> with this edge proxy, for what purpose would you be connecting to it 
>>> in the first place? You should connect directly to your chat provider 
>>> offering the service. Then, if you have a cert - great - authenticate 
>>> to it using mutual TLS.
>>>
>>
>> Well, I might be doing "outbound" with my edge proxy in order to get 
>> some help from it for relaying messaging back through my NAT.
> 
> 
> Why can't you do that with the edge proxy in your home providers network?

You might do that with the edge proxy in your home provider's network. 
But that doesn't mean thet your 3rd-party service provider trusts the 
authentication service provided by your home provider's network.

>> Or it might be a QoS-controlling proxy ala PacketCable. Or a 
>> firewall-control-proxy . . .
> 
> 
> Ah, well QoS control is much better handled through policy server 
> peering than SIP connectivity. And it works for non-SIP things. Why is 
> it that SIP is the vehicle for getting QoS in a visited network? Is this 
> problem not more generic?

Probably. But as I recall, PacketCable does it with SIP.


> 
>>
>> The chat provider might not just be on the other side of my edge 
>> proxy, but on the other side of my home serving proxy too. And the 
>> certificate auth stuff would appear to be able to traverse that proxy 
>> as well.
> 
> 
> I don't follow the picture you have in mind.

Clearly.

Imagine a "home network" consisting of an edge proxy and a home service 
proxy. Imagine a 3rd party service that is NOT in a trust relationship 
with the "home network". How does that 3rd party service validate the 
identity of a user?

Options include:

1) Do a conventional SIP authentication using digest or basic, both of 
which have issues we understand.
2) Trust the P-A-ID (which we've exluded by definition, since we don't 
trust it).
3) Use AIB (which doesn't exist, and we might have just excluded by 
definition as well).
4) Use certificate authentication as proposed.

--
Dean




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From sip-bounces@ietf.org Mon Jul 09 02:35:20 2007
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Jonathan Rosenberg wrote:
> 
> 
> Steve Dotson wrote:
> 
>> Maybe this is naïve, but doesn't this alter the operator trust model?
>>
>>
>> I always envisioned P-A-ID (as an example) as a way for a home
>> network to tell a 3rd party service "trust this one, they are
>> authenticated", as long as the home network and the service had a
>> relationship. Having an edge proxy in a visited domain telling the
>> home network "trust this one" seems to turn things around.
> 
> 
> P-A-ID asserts identity between providers with trust, yes that is true. 
> SIP Identity works with lesser degrees of trust, but if you don't trust 
> the signing domain, of course it doesn't work.
> 
> However, my point is, one of the two following scenarios always apply:
> 
> 1. the edge proxy is trusted by the home proxy, in which case you can 
> use mutual TLS to authenticate the client to the edge proxy, and then 
> P-A-ID or SIP Identity from edge proxy to home proxy, *OR*
> 
> 2. there is no trust at all between visited and home networks. In this 
> case, you shouldn't be connectign to a visited proxy in the first place. 
> COnnect to the home proxy in your home network. Then, case 1 applies again.

Wrongo.

3) There is no trust between the edge proxy and the 
point-of-service-delivery (which MIGHT be a proxy, or might not, and we 
don't know and don't care). But you have to use the edge proxy, because 
it's the only one that can get you out of your little 
nat/firewall/whatever administrative island.

--
Dean


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Rai, Anupam (Anupam) wrote:
> 
> (2) Edge proxy utilizing TLS offloaders.

Also an excellent example, and one I've seen come up in real-world 
planning discussions.

--
Dean




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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Sip] Certificate authentication in SIP
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Jonathan Rosenberg wrote:

>> (2) Edge proxy utilizing TLS
>> offloaders.
> 
> 
> The offloader would need to pass the identity to the server. If it acted 
> as a simple SIP proxy in addition to doing TLS offload, it would work 
> fine. There are other non-standard ways of passing this identity also.
> 

Let's just say that we're aware of commercial TLS concentrators from a 
certain vendor (who might ostensibly be said to fund your participation 
in this forum) that are both very popular and that DO NOT HAVE this 
functionalility and aren't very likely to get it (after all, why would 
you wish to reduce their performance by a factor of 50 or so to start 
parsing SIP messages?). Finding a way other than basic or digest to 
authenticate across them would be very nice, and would be very likely to 
ramp up the deployment of SIP/TLS.

--
Dean


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From sip-bounces@ietf.org Mon Jul 09 03:52:02 2007
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This draft specifies two new response codes:

418 SIPS Not Allowed
419 SIPS Required

With sipsec around the corner shouldn't we make
these headers more generic like we do for other
SIP things?

For example, if someone uses a method we don't allow
then we return "405 Method Not Allowed" and then
list the allowed methods in the Allow header.
And there is a similar solution for SIP extensions
(420 Bad Extension or 421 Extension Required).

So how about:
418 URI Scheme Not Allowed
Allow-URI: sip

419 URI Scheme Required
Require-URI: sips,sipsec

In fact, you probably only need one of them as they are
almost identical in resultant behaviour.

Regards,

Attila



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From sip-bounces@ietf.org Mon Jul 09 08:54:55 2007
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In general this is now in excellent shape. Just some detailed comments.

1. "It also makes normative changes to SIP". If so, then it should state
on the front page that it updates RFC 3261.

2. In 3.1.1 "In this model, only the TLS server provides a certificate
during the
   TLS handshake.  Often, a proxy (which may or may not be the one
   terminating the TLS connection) authenticates the client by using the
   SIP HTTP digest.  In this model, the UA is the TLS client side, and
   the Proxy is the TLS server side."
The words "the Proxy" in the third sentence would appear to refer to "a
proxy" in the second sentence, but that doesn't make sense. Also I think
this is trying to say that the server-provided certificate can only
apply between a UA and a proxy and when the UA is the client, so it
can't apply between proxies and it can't apply when a proxy is a client.
If so, to solve these two issues, perhaps it should say.
"In this model, only the TLS server provides a certificate during the
TLS handshake.  This is applicable only between a UA and a proxy, where
the UA is the TLS client and the proxy is the TLS server, and hence the
UA uses TLS to authenticate the proxy but the proxy does not use TLS to
authenticate the UA. If the proxy needs to authenticate the UA, this can
be achieved by SIP HTTP digest authentication."

3. In 3.1.2 "In this model, both the TLS client and the TLS server
provide a
   certificate in the TLS handshake phase.  When no TLS connection is in
   place, the UAC would therefore take on the role of the TLS Client and
   the UAS would therefore take on the role of the TLS server."
This seems to limit mutual TLS (more correctly TLS mutual
authentication) to use involving a UA, but it also has an important use
between proxies (this is not mentioned until further down). Also the
second sentence seems to suggest a direct TLS connection between UAC and
UAS, which is possible but not the typical case. Where at least one
proxy is involved in the signalling path, if there is not already a TLS
connection in place between the UAC and the first proxy, the UAC would
take on the role of TLS client and the first proxy would take on the
role of TLS server. Likewise, if there is not already a TLS connection
in place between the last proxy and the UAS, the last proxy would take
on the role of the TLS client and the UA would take on the role of the
TLS server. I think this needs to be explained more fully. Perhaps we
need to say something like:
"In this model, both the TLS client and the TLS server provide a
certificate in the TLS handshake phase. When used between a UA and a
proxy (or between two UAs), this implies that a UA must be in possession
of a certificate. When sending a SIP request when there is not already a
suitable TLS connection in place, a UAC takes on the role of TLS client
in establishing a new TLS connection. When establishing a TLS connection
for receipt of a SIP request, a UAS takes on the role of TLS server."

4. "While guessing a SIPS AOR from a known SIP
   AOR and using it to initiate a request is a valid thing to do, doing
   the opposite (i.e., guessing a SIP AOR from a SIPS AOR and using it)
   is not a valid thing to do as it would be a security downgrade."
I would not go as far as saying it is not valid. It is merely
undesirable, but if the caller's UA does not support the SIPS scheme,
that might be a valid way of contacting the desired user.

5. "Although "downgrading" from SIPS to SIP is disallowed,"
As mentioned above, I think it is valid but undesirable for a user to
downgrade by guessing. However, I think what we are talking about here
is downgrading by a proxy, so perhaps that should be made clear by
saying: "Although "downgrading" from SIPS to SIP is disallowed at a
proxy,".

6. In 4.1 "If a UAS does not wish to be reached with a SIPS URI but only
with a
   SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed)."
If the UA does not wish to be reached with a SIPS URI, then surely it
would not register a SIPS contact. If the UA has not registered a SIPS
contact, it should not normally receive a SIPS request. So presumably
this is required only for requests reaching the UA without going through
the domain proxy. Correct? If so, perhaps some explanation is needed.

7. "UAS that
   do not support the SIPS URI scheme altogether "SHOULD reject the
   request with a 416 (Unsupported URI scheme) response" as described in
   [RFC3261]/8.2.2.1."
Similar - should explain how this can occur.

8. In 4.1.1 "If a UA registers with a SIPS Contact header field, the
registrar
   returning a service route [RFC3608] MUST return a service route
   consisting of SIP URIs if the intent of the registrar is to allow
   both SIP and SIPS to be used in requests sent by that client.  If a
   UA registers with a SIPS Contact header field, the registrar
   returning a service route MUST return a service route consisting of
   SIPS URIs if the intent of the registrar is to allow only SIPS URIs
   to be used in requests sent by that UA."
Because 4.1.1 is part of 4.1, which deals only with UAs, then it should
not contain normative requirements on registrars. Presumably the
intention of this is to state that RFC 3608 places such a requirement on
registrars. The language needs changing to reflect this.

9. "However, the registrar MUST treat the SIP and SIPS schemes of the
AOR
   the same way (i.e., it MUST not care if it is SIP or SIPS)."
This is a requirement on the registrar and does not belong in 4.1.

10. "A registrar MUST only accept a binding to a SIPS Contact header
field
   if all the appropriate URIs are of the SIPS scheme, otherwise there
   could be an inadvertent binding of a secure resource (SIPS) to an
   unsecured one (SIP).  This includes the Request-URI, the Contacts and
   all the Path headers, but does not include the From and To headers.
   If the URIs are not of the proper SIPS scheme, the registrar MUST
   reject the REGISTER with a 419 (SIPS Required)."
These are requirements on the registrar and do not belong in 4.1.

11. In 4.2 "It is RECOMMENDED to use an outbound proxy as per the
procedures
   defined in [I-D.ietf-sip-outbound] for supporting UACs that can not
   provide a certificate for establishing a TLS connection (i.e., when
   server-side authentication is used)."
This is a recommendation on a UA, not a proxy, so it does not belong as
a normative statement in 4.2.

12. In 4.3 "A
   redirect server would not be usable if server-side authentication is
   used"
This is incorrect. If a UA has a certificate it can provide TLS
server-side authentication, yet a redirect server could still be used in
this case. I think this is trying to say that a redirect server cannot
be used to redirect to an entity (e.g., a UA) that does not have a
certificate.

13. "When a redirect server receives a request with a SIP Request-URI,
the
   redirect server MAY redirect with a 3XX response to either a SIP or a
   SIPS Contact header field.  If the target UAS had registered
   previously using a SIPS Contact header field, the redirect server
   SHOULD return a SIPS Contact header field if it is in an environment
   where TLS is usable (as described in the previous paragraph)."
This assumes that a redirect server is redirecting directly to a UA,
i.e., to a contact. However, a redirect server can also redirect to
another AoR that resolves to a domain proxy (or another redirect
server). In other words, the redirect server may not have any knowledge
of a "target UAS".

14. In 5.2 "Furthermore, the offending SIPS URI(s) may be in any
   location in the request (e.g., the Request-URI, a Contact header
   field, a Refer-To URI, etc.)."
Shouldn't this say "the offending SIP URI(s)"?

15. In the example in 6.3, proxy A and proxy B violate the
recommendation in 4.2 to route SIP requests over TLS. This doesn't seem
to be a very good example! It would be better to show routing over TLS
from proxy A onwards.

16. "7.  Conclusion"
Most of the content of this section consists of further considerations,
rather than conclusions drawn from previous sections. So perhaps the
title should be "Further considerations".

17. NITS

- In 6.1 "This call flow". There is no call flow - the flow shows only
registration.

- There are instances of a SIP method name (e.g. REGISTER) being used
alone without being followed by "request".

- There are instances of "header" being used instead of "header field".

John






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From sip-bounces@ietf.org Mon Jul 09 12:41:47 2007
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Subject: [Sip] Updated draft-gurbani-sip-domain-certs-06
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On Fri, 2007-06-22 at 16:12 -0500, Vijay K. Gurbani wrote:
> Folks: draft-gurbani-sip-domain-certs-05 has been submitted to
> the IETF archives.  This version includes the guidance we
> got from the pkix WG in Prague.  More specifically, this version
> focuses strictly on:
> 
> - How to use and interpret the SIP identities in a X.509 certificate.
> - How to indicate that this particular certificate is for SIP
>   usage.

We got some excellent late-breaking comments from Stephen Kent, and
reissued this as -06 just ahead of the deadline for Chicago.  Since such
a late submission may take some days to get through the pipeline, you
can get it here now:

http://svn.resiprocate.org/rep/ietf-drafts/gurbani/domain-certs/draft-gurbani-sip-domain-certs-06.txt
http://svn.resiprocate.org/rep/ietf-drafts/gurbani/domain-certs/draft-gurbani-sip-domain-certs-06.html

We are on the current agenda for discussion on Tuesday in Chicago; we
would like to focus discussion on whether or not this is an appropriate
addition as a Working Group item under the existing charter items (and
of course on any problems people may see with the content).

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXecs project coordinator - SIPfoundry http://www.sipfoundry.org/sipXecs
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/



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Subject: [Sip] WGLC for draft-ietf-sip-outbound
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I'd like to start a long working group last call for:

http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt

This WGLC period will run through IETF and hopefully close August 6, 2007.

Please read the draft and send any comments both to the list and direct
to the authors.

We'll also have a few minutes on the agenda in Chicago to discuss the
issues if any pop up.

--
Dean Willis
co-chair, SIP


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From sip-bounces@ietf.org Mon Jul 09 15:58:07 2007
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Subject: [Sip] Proposed: delete Guidelines for use of SIP Security milestones
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We have these two milestones in our charter, and neither Keith nor I are
exactly sure what they mean:

Mar 2007 Guidelines for use of SIP security to WGLC (Informational)
Jun 2007 Guidelines for use of SIP security to IESG (Informational)

Now we all know that there is a substantial need to better document the
usage of and relationships between the many SIP-related security
mechanisms, but I'm not aware of having seen a document posted that is
making any headway on this problem,. and I'm not even sure we have a
consensus on exactly what needs to be done.

Plus, we've definitely missed the milestones.

So I'd like to propose that we delete these milestones for now, and
bring them back only when we have a concrete proposal on the table for
what we think we can actually do (and should do) here.

Opinions?

--
Dean


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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



From sip-bounces@ietf.org Mon Jul 09 16:48:52 2007
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When did they get introduced? Seems odd that we got them added when no 
one is quite sure what they mean.

-Jonathan R.

Dean Willis wrote:
> We have these two milestones in our charter, and neither Keith nor I are
> exactly sure what they mean:
> 
> Mar 2007 Guidelines for use of SIP security to WGLC (Informational)
> Jun 2007 Guidelines for use of SIP security to IESG (Informational)
> 
> Now we all know that there is a substantial need to better document the
> usage of and relationships between the many SIP-related security
> mechanisms, but I'm not aware of having seen a document posted that is
> making any headway on this problem,. and I'm not even sure we have a
> consensus on exactly what needs to be done.
> 
> Plus, we've definitely missed the milestones.
> 
> So I'd like to propose that we delete these milestones for now, and
> bring them back only when we have a concrete proposal on the table for
> what we think we can actually do (and should do) here.
> 
> Opinions?
> 
> --
> Dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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



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Subject: [Sip] Hitchhikers guide to sip: update available
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I submitted an update to the hitchhikers guide:
http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt

Changes:

* connected-identity -> RFC4916
* added record-route-fix and grouped it with fork-loop-fix into a new
item under SIP core called "essential corrections"
* added subnot-etags under event packages
* added sip-sips to core specs

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Dean Willis wrote:
> We have these two milestones in our charter, and neither Keith nor I
> are exactly sure what they mean:
> 
> Mar 2007 Guidelines for use of SIP security to WGLC (Informational) 
> Jun 2007 Guidelines for use of SIP security to IESG (Informational)
> 
> Now we all know that there is a substantial need to better document
> the usage of and relationships between the many SIP-related security 
> mechanisms, but I'm not aware of having seen a document posted that
> is making any headway on this problem,

Well, way back at the 65th IETF in Dallas (March 2006), I had
presented some slides on draft-gurbani-sip-tls-use-00 (slides
archived at http://www3.ietf.org/proceedings/06mar/slides/sip-5.pdf).
Portions of that work got absorbed elsewhere -- for instance,
the URI promotion aspect was then further explored in sip-sips;
the test cases from tls-use-00 made it into sip-sec-flows;
the mutual authentication aspect made it to domain-certs, etc.
tls-use-00 subsequently expired and was not revised, having
served some purpose in life.

> and I'm not even sure we have a consensus on exactly what needs to be
> done

Wouldn't drafts like sec-flows, sip-certs, sip-sips, domain-certs
all lead towards providing guidelines for the use of SIP
security?

> So I'd like to propose that we delete these milestones for now, and 
> bring them back only when we have a concrete proposal on the table
> for what we think we can actually do (and should do) here.

It looks like the drafts and a substantive body of the work in
this area is out there; just needs to be organized for a more
amenable deadline.  But, I may be mistaken...

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://www.alcatel-lucent.com/bell-labs


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From sip-bounces@ietf.org Mon Jul 09 17:20:14 2007
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Fries, Steffen wrote:
 >
 > - 1 Introduction (page 3)
 >
 >    maybe it is worth mentioning in the introduction already that this ID
 > only considers "call by reference" for the assertions and does not provide
 > means for sending the assertion directly.

Sure, we can do that. Though note that perhaps in the longer term it'll depend 
on the resolution to the issue of whether the spec remains strictly 
call-by-reference or evolves to include call-by-value. i.e. see..

[Sip] wrt SIP Saml Review by Jiri Kuthan - "missing scenarios"
http://www1.ietf.org/mail-archive/web/sip/current/msg19793.html


 > - 3.1 SAML assertions (page 7)
 >    "Explaining how the assertion was made, for example. " Sounds like an
 > example would follow,
 >    which does not. I suggest removing "for example"
 >
 > - 4 Specification scope  (page 9), in the discussion paragraph:
 >    "However, The ..." -> should be "However, the ..."
 >    "participating entites ..." -> "participating entities ..."
 >
 > - 5. Emplying SAML in SIP (page 11)
 >   "seperatey" -> "separately"
 >
 > - 6.1.2 Profile Overview (page 14)
 >   "Although this profile is overview is cast in terms of a SIP INVITE
 > ..." There is something wrong with this sentence
 >
 > - 6.1.3.2 (page 19)
 >   procedes -> proceeds

I've entered all these editorial comments into the sip-saml tracker (under "bug")..

http://www.tschofenig.priv.at:8080/saml-sip/issue?@sort=-activity&@group=priority&@filter=status&@columns=id,activity,title,creator,assignedto,status&status=-1,1,2,3,4,5,6,7


thanks for your review,

=JeffH





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From sip-bounces@ietf.org Mon Jul 09 17:47:49 2007
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 > -------- Original Message --------
 > Subject: [Sip] SIP Saml Review by Marcos Dytz
 > Date: Mon, 18 Jun 2007 10:39:20 +0200
 > From: Tschofenig, Hannes <hannes.tschofenig@nsn.com>
 > To: SIP IETF <sip@ietf.org>
 > CC: ext Marcos Dytz <marcos.dytz@gmail.com>
 >
 > 1) ...  (or, at
 > least, specifying that it can be used to convey attributes, query for
 > information, etc).

yes, we can attempt to clarify this in the draft.


 > 2) The SAML assertion profile sounds weird, although I never completely
 > had a complete grasp of its meaning.

You're referring to section "6.1.4.  Assertion Profile Description" of 
-sip-saml-02, yes?

 > I understand that it is connected
 > with the implementation, but it never actually had any particular use to
 > me (or I never gave it much attention). I rather see it as details that
 > developers had to apply in order to have the implementation working and
 > must be added to the specification in order to avoid interoperability
 > issues.

I called it an "assertion profile" because it "profiles" SAML assertions for 
use in this particular context. SAML assertions are explicitly tailorable, so 
each SAML profile needs to do this as well as specify how they are to be 
conveyed. Since they are security tokens, explicitly defining their content and 
the semantics thereof is important.



 > 3) I believe it is worth mentioning that one invalid condition locks the
 > whole assertion.

by "locks" you mean "invalidates"?  Sure we can try to clarify this, but in 
section "6.1.5.  Assertion Verification", step 11, we do note that the validity 
period must be validated -- tho yes we should perhaps explicitly state that an 
  assertion that fails any of the 11 steps MUST be deemed invalid (good catch).


 > 4) I also believe that is worth mentioning that the Identity-Info is
 > connected with CERTS and might go through deployment issues.

This is tied to our perhaps thorniest open issue, which is discussed here..

[Sip] wrt RFC4474 Identity-Info header field referent issue (was: SIP SAML 
Review by Richard Barnes)
http://www1.ietf.org/mail-archive/web/sip/current/msg19686.html

http://www.tschofenig.priv.at:8080/saml-sip/issue12

So, yes, we need to clarify appropriately once we deal with the issue.


 > 5) Is there something missing on 7.1? Sounded incomplete to me.

AFAIK it is complete in a technical sense. I believe some other reviewers had 
more detailed comments on it, so will likely be looking into it in any case.


 > 7) The part of the AS as an authenticator and proxy could be further
 > clarified.

which section/paragraph are you referring to?

thanks for your review,

=JeffH




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From sip-bounces@ietf.org Mon Jul 09 18:07:14 2007
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Hannes Tschofenig wrote:
 > Here is a review by Andreas Pashalidis:
 >
 > ----------------------
 >
 > Here are a few small comments:
 >
 > "trait-based" authentication sounds a bit unconventional. Cannot you
 > just talk about "attribute-based" authorization? That would be better
 > aligned with SAML terminology.

Since this spec is being concocted in the context of the SIP working group, I 
felt using SIP terminology was/es the way to go, see RFC4484. Though, we could 
add an entry(ies) to the "2. terminology" section to help clarify items such as 
this.


 >> From the description in section 5 and figure 1, it is not always clear
 > if you talk about an "authentication assertion" or an "attribute
 > assertion"

I kept this deliberatively vague because section 5 is very high level. We could 
perhaps add a sentence to the effect that a SAML assertion can be used to 
assert attributes of a subject's authentication event and/or profile attributes.


 > Figure 2, step 5: is it possible to have multiple attribute statements
 > in the response? (for example, if possession of multiple attributes is
 > required?)

yes. Also note that a given <saml:AttributeStatement> may assert any number of 
attributes.


 > Section 6.1.4.1.4: what does it mean if there is no attribute
 > statement?

good point, the "MAY" in the first sentence arguably should be a "SHOULD".

However, some implementation & deployment might be conveying all the 
information they need with other subelements of the SAML assertion, and so they 
get all the semantics out of the assertion that they need without a 
<saml:AttributeStatement>.

 > is it an authentication assertion then?

No, not unless it explicitly contains an <saml:AuthnStatement>.


 > Section 9.2: it would be nice to have some exaplanation there, without
 > having to refer to a different spec/document.

ok, will think about what we can add there.

thanks for your review,

=JeffH



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Dan,

I just took a look at this. I have a few questions:

1) Because you only sign the selected attributes from the SDP, the rest 
is not secured. That seems like an acceptable restriction if you assume 
there will be a media-terminating SBC in the path.

But what about other body parts? I gather they won't be signed either. 
That seems like a potential problem as we start to identify need for 
various extra body parts.

2) What about "offerless" initial INVITEs?

3) What about messages that don't use SDP at all?

	Thanks,
	Paul






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 > -------- Original Message --------
 > Subject: [Sip] SIP SAML Review by Richard Barnes
 > Date: Wed, 20 Jun 2007 23:44:26 -0400
 > From: Richard Barnes <rbarnes@bbn.com>
 > To: IETF SIP List <sip@ietf.org>, 	Hannes Tschofenig <Hannes.Tschofenig@gmx.net>
 >
 >
 > Figure 2:
 > In the INVITE messages, the AoR in the From: header should be
 > "sip:alice@foo.com"

ok.


 > Section 6.1.2, Step 3:
 > The AS is supposed to insert "attributes required by Bob's domain".  How
 > is it supposed to know which attributes these are?  This is answered by
 > the following paragraph, which should be moved up to here.

ok.


 > Section 6.1.2, Step 3:
 > The AS constructs a "HTTP-based SAML URI reference" -- shouldn't this be
 > a reference according to the binding defined in this document, instead
 > of the original (SAMLv2) reference standard?

nominally yes, will investigate.


 > Section 6.1.3.3, third paragraph:
 > This paragraph gives the option of HTTPS or HTTP, but doesn't specify
 > that one of the two MUST be used.  Suggested text: "The AS MUST
 > construct either an HTTP or HTTPS URI per Section "3.7.5.1 URI Syntax"
 > of [OASIS.saml-bindings-2.0-os].  It is RECOMMENDED that the AS
 > construct an HTTPS URI."

good catch.


 > Section 6.1.3.3, fourth paragraph:
 > This is not technically correct; the URI should be the absoluteURI
 > portion of the Identity-Info header value.

another good catch.


 > Section 6.1.4:
 > Through out this section, situations with exceptions are presented as
 > "SHOULD ... MAY ...".  As I read it, this creates a situation without a
 > mandatory default behavior; I would prefer if these were worded as
 > MUSTs, with MAYs allowed under specified circumstances.

will re-think with eye towards mandatory default behavior.


 > Section 6.1.4.1.3:
 > The SHOULDs in this section should be should be MUSTs.

nominally agree.


 > Section 6.1.4.1.4, second paragraph:
 > This needs to say that the attribute-value pairs MUST pertain to the
 > subject of the assertion (a concept which needs to be tied to a SIP
 > entity elsewhere in the document).  This may be the entity identified in
 > the From header, or another identified entity.

nominally agree.


 > Section 7:
 > This binding doesn't really have anything to do with SIP.

agreed.

 > Could we just
 > exclude SIP from the name?

sure.


 > Maybe this could be the IETF standard URI
 > binding of SAML.

that's an interesting thought, tho I wonder if it is worth the effort to try to 
do as an I-D on its own -- it's so small to specify that it can easily be 
included by-value in other specs, but of course then there might be skew, but 
would that in actuality be a problem, ... etc. And more precisely it's an 
http-uri-based one. there could be other needs for schlepping saml assertions 
via http in other fashions.


thanks for your insightful review.

=JeffH




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 > -------- Original Message --------
 > Subject: 	[Sip] Comments draft-ietf-sip-saml-02
 > Date: 	Thu, 21 Jun 2007 16:47:46 +0200
 > From: 	Willekens, Marc <marc.willekens@nsn.com>
 > To: 	<sip@ietf.org>
 >
 >
 > [Page 3], Introduction
 >
 > Is this document only about assertion for the authorization or also for
 > the authentication?

authorization.


 > [Page 12]
 >
 > Domain name for Alice: example.com or foo.com?? (same comment for other
 > figures)

oops, good catch. foo.com.


 > Authentication service or Assertion service or both?

authentication service -- the term is from rfc4474.


 > [Page 13], 6.1. Assertion Fetch profile
 >
 > Is this the only planned profile or will other profiles be defined and
 > added later?

My personal present thinking is that there are other possible SIP SAML 
profiles, e.g. see the appendix in -sip-saml-02 of other use cases, but that 
they should likely be specified separately in their own drafts.

 > Will there also be a push based model? Will it then be
 > "pushed" as part of the SIP message?

This is something for the WG to think about. One _can_, in theory, have the UAC 
assert some information and attach a SAML assertion to a SIP request (e.g. 
Invite) body making such a statement. Thus this approach could only be used by 
UACs because (1) only UACs may place data in the SIP request body, and (2) 
there are vociferous objections in the SIP WG to schemes whereby SIP 
intermediaries may place BLOBs (binary large objects) in SIP header fields. See 
also..

[Sip] wrt SIP Saml Review by Jiri Kuthan - "missing scenarios"
http://www1.ietf.org/mail-archive/web/sip/current/msg19793.html


We could specify this for UACs-only, and the other http-uri-based fetch 
approach for, as it is currently, intermediates or UACs.

However, before we go to the effort, what's the in-actuality demand for doing 
so? Has anyone implemented the rfc4474 authentication service role in the UAC 
and is it being used?


 > [Page 13], 6.1.1. URN
 >
 > What's the current consensus for the urn?

No one has objected to what is in the spec, although no one has commented 
specifically on it as yet, either.



 > [Page 17], 6.1.3.
 >
 > ...sender and Authentication service (AS) may be separate or... ???
 >
 > How will the AS and sender be combined?

See the last paragraph on page 5 of rfc4474..

   "This specification allows either a user agent or a proxy server to
    provide identity services and to verify identities."



 > [Page 22], 6.1.4.1.4.
 >
 > AttributeStatement is indicated. But what about Authentication and
 > Authorization Decision statement?

This particular SIP SAML Profile doesn't specify use of those statements, but 
they are not explicitly excluded either. Conceivably someone could cook up a 
use case for them and make use of it in a site-specific manner, or we might 
find use case and then can specify their use more easily.

If we do want to explicitly do this we should perhaps add that such statements 
MAY be present, but their meaning is defined outside of this profile.


 > [Page31]
 >
 > This RFC draft talks about assertions. But how can an identity be
 > asserted only by considering a device, e.g. in case a device is stolen
 > or temporary used by someone else? At VON 2000, H. Schulzrinne has
 > already indicated that something as fingerprint has to be added for a
 > device with a changing set of owners.

Although a valid thought, that being the question of the nature of the actual 
authn mechanism between the UAC and the registration service, it is out of 
scope of this specification. In fact, this draft nowhere states that the 
subject's identity is asserted only by the device (UAC).


thanks for your review,

=JeffH









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 > ------- Original Message --------
 > Subject: [Sip] SIP SAML Draft Review by Sebastian Felis
 > Date: Wed, 27 Jun 2007 12:02:48 +0200
 > From: Tschofenig, Hannes <hannes.tschofenig@nsn.com>
 > To: SIP IETF <sip@ietf.org>
 >
 >
 > - Another topic is the usage of the term "Authentication Server" which
 > has the acronym AS. Since the term is originating from the trait-based
 > authorization draft,

well, it's actually from rfc4474 "SIP Identity".


 > this term should be fine. But from the IMS
 > perspective, the acronym AS is "Application Server". By today, the SIP
 > (p)rotocol is more and more associated with IMS and touches the terms of
 > IMS. Further, the term Authentication Server is not a common term in
 > SAML. So I propose to use another term, which is more SIP/IMS and/or
 > SAML conform, e.g. Identity Provider in case of SAML.

Well, this draft spec is being concocted in the context of the SIP WG, and so I 
am trying to have the terminology be as SIP-driven as possible.

We can note, though, in the terminology section, that "AS" is "app server" in 
the IMS world in order to try to head off confusion.

thanks for your review,

=JeffH





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From sip-bounces@ietf.org Mon Jul 09 20:39:21 2007
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Vijay K. Gurbani wrote:

> Wouldn't drafts like sec-flows, sip-certs, sip-sips, domain-certs
> all lead towards providing guidelines for the use of SIP
> security?

Certainly, and these are all both within the charter of the working 
group, and individually milestoned when the WG develops a consensus to 
deliver them.

>> So I'd like to propose that we delete these milestones for now, and 
>> bring them back only when we have a concrete proposal on the table
>> for what we think we can actually do (and should do) here.
> 
> 
> It looks like the drafts and a substantive body of the work in
> this area is out there; just needs to be organized for a more
> amenable deadline.  But, I may be mistaken...

Otherwise said, you think the milestones in question were essentially 
made redundandt by the adoption of the more-focused drafts? If so, then 
we can just delete the milestones in question, as we know we can add new 
milestones for new "more focused" drafts under our existing charter.

Or do we need something like a meta or framework document that brings 
all these pieces together into a more coherent structure?


--
Dean




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At 07:42 PM 7/9/2007, Dean Willis wrote:
>Vijay K. Gurbani wrote:
>
>>Wouldn't drafts like sec-flows, sip-certs, sip-sips, domain-certs
>>all lead towards providing guidelines for the use of SIP
>>security?
>
>Certainly, and these are all both within the charter of the working 
>group, and individually milestoned when the WG develops a consensus 
>to deliver them.
>
>>>So I'd like to propose that we delete these milestones for now, 
>>>and bring them back only when we have a concrete proposal on the table
>>>for what we think we can actually do (and should do) here.
>>
>>It looks like the drafts and a substantive body of the work in
>>this area is out there; just needs to be organized for a more
>>amenable deadline.  But, I may be mistaken...
>
>Otherwise said, you think the milestones in question were 
>essentially made redundandt by the adoption of the more-focused 
>drafts? If so, then we can just delete the milestones in question, 
>as we know we can add new milestones for new "more focused" drafts 
>under our existing charter.
>
>Or do we need something like a meta or framework document that 
>brings all these pieces together into a more coherent structure?

this meta framework was discussed in Dallas and it was basically, 
without structure at the time - suggested that SIP ought to get its 
act together and make SIP Security make sense to everyone else.

(Dean, you were too busy 'packing heat'  to notice  ;-)

(everyone else might have been packing their canoe from that storm  :-D

I'll suggest, if possible, there is a call for volunteers to write 
such a doc, including what's advanced since 65. Therefore I don't 
want that milestone removed until we can't find someone to write 
it.  Moving the dates out a year seems prudent though.



>--
>Dean
>
>
>
>
>_______________________________________________
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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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Rohan Mahy wrote:
> Hi Fredrik,
> 
> 1) In this version I added text in the current Security Consideration
> to help clarify this.  See the diffs in Section 13 here:
> 
> http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-outbound.diff.html 

Thanks for pointing that out.

> I thought I already sent you a message about this topic, but in any
> case I will post my response here.  The goal is to make sure that *no
> new attacks* are made possible by the introduction of this extension.

Ok. Adding complexity can easily lessen security though. I don't agree 
to the principle of adding something meant for security when you're not 
sure it actually helps in any case [1], so unless you figure out _why_ 
the draft needs to say these things (and document that, please) I would 
opt for removing it from the draft.

I will however stop nagging about this now, since you are the document 
editor and not me.

/Fredrik

[1] like an unnamed Swedish bank allegedly having seven identically 
configured firewalls connected serially between the Internet and their 
internal network


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From sip-bounces@ietf.org Tue Jul 10 05:14:26 2007
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A reminder:

When making comments, please clearly identify:

-	the section being commented upon, and therefore impacted by any =
resolution (extracted text is also useful)
-	the level of the comment (NIT, minor technical, major technical)
-	the comment itself
-	any suggested resolution

We will tabulate any reviews according to the above.

Regards

Keith
=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Monday, July 09, 2007 8:41 PM
> To: sip@ietf.org
> Cc: Cullen Jennings; DRAGE, Keith (Keith); rohan.mahy@gmail.com
> Subject: WGLC for draft-ietf-sip-outbound
>=20
> I'd like to start a long working group last call for:
>=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
>=20
> This WGLC period will run through IETF and hopefully close=20
> August 6, 2007.
>=20
> Please read the draft and send any comments both to the list=20
> and direct to the authors.
>=20
> We'll also have a few minutes on the agenda in Chicago to=20
> discuss the issues if any pop up.
>=20
> --
> Dean Willis
> co-chair, SIP
>=20


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From sip-bounces@ietf.org Tue Jul 10 05:21:18 2007
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Hi

Which is the appropriate 4xx message that can be used to inform user of
an existing subscription?=20

Consider the following use case: User A sends a subscribe message with a
particular event and gets subscribed to the service. Now at a later
point of time the user is not sure if he has subscribed for a particular
service or not. So he might send subscribe again. So is there a way to
inform the user that the subscription for that service for the user
already exists?

Thanks=20


Regards
Ranjit



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From sip-bounces@ietf.org Tue Jul 10 06:45:46 2007
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Subject: RE: [Sip] Changes to the INVITE transaction to address bug 769
Date: Tue, 10 Jul 2007 12:45:31 +0200
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From: "Bohaty David" <David.Bohaty@sitronicsts.com>
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Hello,
I have 2 more thoughts on the draft:

First comment is related to the state machine for INVITE server =
transaction in which I would say is missing treatment for the transport =
error in the "Accepted" state. On transport error there could be new =
transition from "Accepted" to "Terminated" with label "Transport Error =
Inform TU".

Second comment is related to treatment of responses to INVITEs forked by =
a proxy further downstream. In that case multiple dialogs can be =
established and we should not just accept from or deliver to TU =
different 200-299 responses after the first one have been received, but =
also 101-199 which can related to the other early dialogs and depending =
on the timing can arrive even after the 200-299 for the first dialog. So =
for client transaction in the "Completed" state the label "2xx 2xx to =
TU" could be changed to "101-299 101-299 to TU". For the server =
transaction in the "Accepted" state the label could be changed from "2xx =
from TU send response" to "101-299 from TU send response". The TU should =
submit 101-199 to the server transaction in the "Accepted" state only if =
it is proxy and only when it is related to different dialog than the =
already confirmed one (if it is dialog-stateful proxy).


David Bohat=FD
System Architect

SITRONICS Telecom Solutions, Czech Republic a.s.=20
Tel.: +420 226 532 578, Fax: +420 241 480 899
BB Centrum - Beta, Vyskocilova 1481/4, 140 00  Praha 4, Czech Republic
www.sitronicsts.com

DISCLAIMER
This e-mail may be privileged and/or confidential, and the sender does =
not waive any related rights and obligations. Any distribution, use or =
copying of this e-mail or the information it contains by other than an =
intended recipient is unauthorized. If you received this e-mail in =
error, please advise me (by return e-mail or otherwise) immediately

-----Original Message-----
From: Robert Sparks [mailto:rjsparks@estacado.net]=20
Sent: 28 June 2007 04:35
To: SIP IETF
Subject: [Sip] Changes to the INVITE transaction to address bug 769

Everyone -

I've put together a proposal for addressing the spec bug captured at
http://bugs.sipit.net/show_bug.cgi?id=3D769

Please read through
http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
and comment.

RjS


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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 =
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Subject: Re: [Sip] 4xx message for informing user of an existing subsription
References: <46928F5F.3080502@softarmor.com>	<5D1A7985295922448D5550C94DE29180013F94BC@DEEXC1U01.de.lucent.com>
	<750BBC72E178114F9DC4872EBFF29A5B04CD8662@ZMY16EXM66.ds.mot.com>
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There is no error response for this, because it isn't an error. It is 
quite legal to establish multiple subscriptions for the same event. 
While in some cases this might be redundant and silly, it is difficult 
or impossible for the notifier to distinguish the useful cases from the 
silly ones.

	Paul

Avasarala Ranjit-A20990 wrote:
> Hi
> 
> Which is the appropriate 4xx message that can be used to inform user of
> an existing subscription? 
> 
> Consider the following use case: User A sends a subscribe message with a
> particular event and gets subscribed to the service. Now at a later
> point of time the user is not sure if he has subscribed for a particular
> service or not. So he might send subscribe again. So is there a way to
> inform the user that the subscription for that service for the user
> already exists?
> 
> Thanks 
> 
> 
> Regards
> Ranjit
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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From sip-bounces@ietf.org Tue Jul 10 10:48:17 2007
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From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Dean Willis wrote:
vkg> It looks like the drafts and a substantive body of the work in
vkg> this area is out there; just needs to be organized for a more
vkg> amenable deadline.  But, I may be mistaken...
> 
> Otherwise said, you think the milestones in question were essentially 
> made redundandt by the adoption of the more-focused drafts? 

No; I thought that the focused drafts were *leading* to the milestone.
But this may be my mis-interpretation of how a WG is chartered.  Is
it the case that it is one draft per milestone?  Or could a series
of drafts satisfy one milestone?

If the latter, then sec-flows, sip-sips, sip-certs, domain-certs
all go towards fulfilling the to fulfil a general milestone that
charters the WG to develop "Guidelines for the use of SIP
security."

> Or do we need something like a meta or framework document that brings 
> all these pieces together into a more coherent structure?

Doesn't S15 of sip-hitchhikers-guide-03
(http://tools.ietf.org/html/draft-ietf-sip-hitchhikers-guide-03#section-15)
come close to providing some structure to the various security-
related drafts?  Plus, if sip-hitchhikers-guide is the last document
that the WG produces, then it will keep up -- thanks to Jonathan --
with newer technologies (SAML), protocols (DTLS) and documents as
they are adopted by the WG.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://www.alcatel-lucent.com/bell-labs


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=20
Folks,
=20
I am seeking why there's a fundamental transition from strict-routing
"RFC2543" to loose-routing "RFC3261".
=20
I find some text in draft-sparks-sip-looseroute-00
=20
   "Several problems with
   the strict routing mechanisms already available in SIP were
   identified, including the endpoint's lack of complete knowledge of
   what that set of proxies should be and the potential loss of
   information about the desired target due to the strict replacement of
   the Request-URI."
=20
But there's no detail description on these problems, could some one
provide me with some more info (link, email, draft, memo) pls?

Sorry if question has been raised before, however, I do not find the
explicit answer.
=20
Thanks
=20
Regards,
Rockson Li (Zhengyu Li)

Software Engineer
Cisco R&D Center, Shanghai, PRC
Phone(Office) : (8621) 24057003
Email: zhengyli@cisco.com


_______________________________________________
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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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Folks,
=20
I am seeking why there's a fundamental transition from strict-routing
"RFC2543" to loose-routing "RFC3261".
=20
I find some text in draft-sparks-sip-looseroute-00
=20
   "Several problems with
   the strict routing mechanisms already available in SIP were
   identified, including the endpoint's lack of complete knowledge of
   what that set of proxies should be and the potential loss of
   information about the desired target due to the strict replacement of
   the Request-URI."
=20
But there's no detail description on these problems, could some one
provide me with some more info (link, email, draft, memo) pls?
=20
thanks
=20
Regards,
Rockson Li (Zhengyu Li)

Software Engineer
Cisco R&D Center, Shanghai, PRC
Phone(Office) : (8621) 24057003
Email: zhengyli@cisco.com

=20

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<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007>Folks,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D833424804-07072007>I am =
seeking why=20
there's a fundamental transition from strict-routing "RFC2543" to =
loose-routing=20
"RFC3261".</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D833424804-07072007>I find =
some text in=20
draft-sparks-sip-looseroute-00</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D833424804-07072007>&nbsp;&nbsp;=20
"Several problems with<BR>&nbsp;&nbsp; the strict routing mechanisms =
already=20
available in SIP were<BR>&nbsp;&nbsp; identified, including the =
endpoint's lack=20
of complete knowledge of<BR>&nbsp;&nbsp; what that set of proxies should =
be and=20
the potential loss of<BR>&nbsp;&nbsp; information about the desired =
target due=20
to the strict replacement of<BR>&nbsp;&nbsp; the=20
Request-URI."</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D833424804-07072007>But =
there's no=20
detail description on these problems, could some one provide me with =
some more=20
info (link, email, draft, memo) pls?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D833424804-07072007>thanks</SPAN></FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Regards,<BR>Rockson Li (Zhengyu Li)<BR><BR>Software=20
Engineer<BR>Cisco R&amp;D Center, Shanghai, PRC<BR>Phone(Office) : =
(8621)=20
24057003<BR>Email: zhengyli@cisco.com<BR></DIV></FONT>
<DIV>&nbsp;</DIV></BODY></HTML>

------_=_NextPart_001_01C7C053.067AA7B9--




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--===============0000049421==--






From sip-bounces@ietf.org Tue Jul 10 13:03:40 2007
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Subject: RE: [Sip] Open issue in the hitchhikers guide - seriously! Relation
	to essential corrections
Date: Tue, 10 Jul 2007 19:03:26 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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Your proposal as to what goes into the hitchhiker's guide sounds =
sensible to me.=20

I will take the comment about the file nameing convention into the next =
version of my essential-corrections draft. Proposals for a naming =
convention welcome; if no alternatives I will use something based on =
what Jonathan suggests below. I see only two issues:

-	we seem to be unable to decide what is a fix and what is not. If we =
make a decision that something is a fix after the -00 version, we need =
to decide whether the continuity of the existing nameing has priority or =
the convention of having -fix in the name.

-	there are already a number of candidate drafts, which obviously do not =
yet follow this convention. Again we need to decide whether to keep the =
continuity of the existing naming or not.

In regard to the fork-loop-fix, this needs to be rediscussed in the WG =
and has agenda time at the forthcoming meeting.

We are looking to get a website or WIKI up that clearly indicates what =
is in progress.

Regards

Keith

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Thursday, July 05, 2007 9:18 PM
> To: IETF SIP List
> Subject: [Sip] Open issue in the hitchhikers guide -=20
> seriously! Relation to essential corrections
>=20
> I was working on my regular update to the sip hitchhikers guide:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers
> -guide-02.txt
>=20
> and ran into an issue. We are now pursuing the SIP essential=20
> corrections
> process:
> http://www.ietf.org/internet-drafts/draft-drage-sip-essential-
> correction-01.txt
>=20
> which has begun to produce internet drafts, that are=20
> draft-ietf-sip-XXX, which will be tracked and produced by the=20
> working group. The question is, do these drafts get included=20
> into the hitchhikers guide or not. My view on this, is NO,=20
> they do not. The reason is that these drafts will never reach=20
> RFC, only the collected essential correction RFC. So, once=20
> that first RFC is produced, the hitchhikers guide would=20
> reference that, and that would encompass everything.
>=20
> So, in the interim, what I'd propose is that the running I-D=20
> version of hitchhikers guide have an "essential corrections"=20
> draft, which in fact references all of the I-Ds currently in=20
> line for essential correction.=20
> Unfortunately, its not clear which drafts those are.=20
> record-route-fix, yes. What about loop-fix? Is that going in?=20
> It would be convenient to have a way of clearly knowing what=20
> drafts are going to be for essential corrections. Is the rule:
>=20
> draft-ietf-sip-*-fix
>=20
> are drafts targeted for essential correction? If so, I think=20
> that should be clearly stated in the=20
> drage-essential-corrections draft.
>=20
> Thanks,
> Jonathan R.
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Tue Jul 10 14:40:42 2007
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Subject: RE: [Sip] I-D Action:draft-ietf-sip-hitchhikers-guide-03.txt 
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Two comments on the sigcomp part of this document.

While Sigcomp is mentioned in passing, RFC 3320 is referenced and this
has been updated by RFC 4896. The ROHC people also consider RFC 4077 an
integral part of the Sigcomp package and RFC 4896 assumes this.

Additionally draft-ietf-rohc-sigcomp-sip is currently with IESG and
this, among other things, defines a SIP Via header parameter and this
should probably be referenced as a SIP extension.=20

Regards

Keith

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Friday, July 06, 2007 3:10 AM
> To: i-d-announce@ietf.org
> Cc: sip@ietf.org
> Subject: [Sip] I-D Action:draft-ietf-sip-hitchhikers-guide-03.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> This draft is a work item of the Session Initiation Protocol=20
> Working Group of the IETF.
>=20
>                                                              =20
>                             =20
>         Title           : A Hitchhiker's Guide to the Session=20
> Initiation Protocol (SIP)
>         Author(s)       : J. Rosenberg
>         Filename        : draft-ietf-sip-hitchhikers-guide-03.txt
>         Pages           : 32
>         Date            : 2007-07-05
>                                                              =20
>                             =20
> The Session Initiation Protocol (SIP) is the subject of=20
> numerous specifications that have been produced by the IETF. =20
> It can be difficult to locate the right document, or even to=20
> determine the set of Request for Comments (RFC) about SIP. =20
> This specification serves as a guide to the SIP RFC series. =20
> It lists the specifications under the SIP umbrella, briefly=20
> summarizes each, and groups them into categories.
>                                                              =20
>                             =20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers
> -guide-03.txt
>                                                              =20
>                             =20
> To remove yourself from the I-D Announcement list, send a=20
> message to i-d-announce-request@ietf.org with the word=20
> unsubscribe in the body of the message.
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce
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>                                                              =20
>                             =20
>                                                              =20
>                             =20
> Internet-Drafts are also available by anonymous FTP. Login=20
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> address. After logging in, type "cd internet-drafts" and then
>         "get draft-ietf-sip-hitchhikers-guide-03.txt".
>                                                              =20
>                             =20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html or=20
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>                                                              =20
>                             =20
>                                                              =20
>                             =20
> Internet-Drafts can also be obtained by e-mail.
>                                                              =20
>                             =20
> Send a message to:
>         mailserv@ietf.org.
> In the body type:
>         "FILE=20
> /internet-drafts/draft-ietf-sip-hitchhikers-guide-03.txt".
>                                                              =20
>                             =20
> NOTE:   The mail server at ietf.org can return the document in
>         MIME-encoded form by using the "mpack" utility.  To use this
>         feature, insert the command "ENCODING mime" before the "FILE"
>         command.  To decode the response(s), you will need=20
> "munpack" or
>         a MIME-compliant mail reader.  Different=20
> MIME-compliant mail readers
>         exhibit different behavior, especially when dealing with
>         "multipart" MIME messages (i.e. documents which have=20
> been split
>         up into multiple messages), so check your local=20
> documentation on
>         how to manipulate these messages.
>                                                              =20
>                             =20
>                                                              =20
>                             =20
> Below is the data which will enable a MIME compliant mail=20
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> version of the Internet-Draft.
>                                                              =20
>                             =20
>=20


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From: "Dan Wing" <dwing@cisco.com>
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Subject: RE: [Sip] SIP Identity using Media Path
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Paul Kyzivat wrote:
> Dan,
> 
> I just took a look at this.

Thanks.

> I have a few questions:
> 
> 1) Because you only sign the selected attributes from the 
> SDP, the rest is not secured. That seems like an acceptable 
> restriction if you assume there will be a media-terminating 
> SBC in the path.
> 
> But what about other body parts? I gather they won't be 
> signed either. 

Right.

> That seems like a potential problem as we start to identify 
> need for various extra body parts.

That's a good point, which I hadn't considered.  

The existing SIP-Identity (RFC4474) could be used to sign such
SIP requests, so long as they don't contain any SDP bodyparts
(if they contain SDP bodyparts and RFC4474 signs them, you're
back at the problems with RFC4474 signatures being broken by
SBCs and B2BUAs).  So, this doesn't seem to be ideal.

What I would suggest, and can write up if it sounds sensical,
is to sign each individual non-SDP MIME part.  This would be
accomplished by creating two new SIP headers, "Identity-part",
which contains the Content-IDs of the non-SDP MIME parts
that are signed, and Identity-Part-Signature, which contains
the signature over those parts.

For example, if a SIP UA sent an Invite like this:

  INVITE ...
  From: Alice@example.com
  To: john@example.net
  Content-Type: multipart/mixed; boundary=X
  ...

  --X
  Content-Type: application/sdp
  
  v=0
  o=- 6418913922105372816 2105372818 IN IP4 192.0.2.1
  s=example2
  c=IN IP4 192.0.2.1
  t=0 0
  m=audio 54113 RTP/SAVP 0
  a=setup:active
  a=connection:new
  a=fingerprint:SHA-1
     4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB

  --X
  Content-Type: text/plain

  John, let's talk about the upcoming press release.

  --X--

The authentication service would add the SIP headers described in the
existing -sip-identity-media-00:


  Identity-Fingerprints:
      4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
  Identity-Media:
     "ZYNBbHC00VMZr2kZt6VmCvPonWJMGvQTBDqghoWeLxJfzB2a1pxAr3VgrB0SsSAa
      ifsRdiOPoQZYOy2wrVghuhcsMbHWUSFxI6p6q5TOQXHMmz6uEo3svJsSH49thyGn
      FVcnyaZ++yRlBYYQTLqWzJ+KVhPKbfU/pryhVn9Yc6U="

and I am proposing it would add the missing Content-Id headers (so that each
MIME part can be clearly identified) and two new SIP headers (Identity-Part,
Identity-Part-Signature).  The 
Identity-Part-Signature contains a signature over all of the MIME bodyparts
listed in the Identity-Part header (shown below with "$" on the left-most
column):

  INVITE ...
  From: Alice@example.com
  To: john@example.net
  Content-Type: multipart/mixed; boundary=X
  Identity-Fingerprints:
      4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
  Identity-Media:
       "ZYNBbHC00VMZr2kZt6VmCvPonWJMGvQTBDqghoWeLxJfzB2a1pxAr3VgrB0SsSAa
        ifsRdiOPoQZYOy2wrVghuhcsMbHWUSFxI6p6q5TOQXHMmz6uEo3svJsSH49thyGn
        FVcnyaZ++yRlBYYQTLqWzJ+KVhPKbfU/pryhVn9Yc6U="
  Identity-Part: ab111
  Identity-Part-Signature:
       "ncEjfecq2jefmceejcech3233rjfejncecjkeu3fjefje93jnakjjqknenceuHEc
        jfeqcnekueri283jfkjfcnencehutiujfe9383mjfejkequie839283jnecneqce
        q11jfjfkcaqncerEJCEJCMEQMCECEJKEFJEKFjejffke"
  ...

  --X
  Content-Type: application/sdp
  Content-Id: ab000
  
  v=0
  o=- 6418913922105372816 2105372818 IN IP4 192.0.2.1
  s=example2
  c=IN IP4 192.0.2.1
  t=0 0
  m=audio 54113 RTP/SAVP 0
  a=setup:active
  a=connection:new
  a=fingerprint:SHA-1
     4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB

  --X
$ Content-Type: text/plain
$ Content-Id: ab111
$
$ John, let's talk about the upcoming press release.
$
  --X--


Thoughts on this approach?


We might also consider using the authentication service's S/MIME keys (not
the UA's S/MIME keys), which may well be cleaner.


> 2) What about "offerless" initial INVITEs?
> 
> 3) What about messages that don't use SDP at all?

Both of these situations would be handled in the same way RFC4474
(SIP-Identity) handles them:

For offerless invites, you would rely on Connected-Identity to provide the
assertion of the initiator's identity.

For messages with no SDP, you can sign non-SDP bodyparts (as I describe
above).  If the message has no body to sign, there isn't any magic we can do
using the media path to authenticate the message came from the claimed peer
-- because of course there is no media path.  Both RFC4474 and RFC3893 (AIB)
share this same problem.  When there is no body, all you can do is sign
headers (including the Date: header) in an effort to prevent replay attacks.
This is what SIP-Identity-Media, RFC4474, and RFC3893 (AIB) all do.

-d


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From sip-bounces@ietf.org Tue Jul 10 17:11:52 2007
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Date: Tue, 10 Jul 2007 17:11:21 -0400
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Folks,

I've submitted a new version of the "Essential Correction to the Session 
Initiation Protocol (SIP) 503 (Service Unavailable) Response":
http://www.ietf.org/internet-drafts/draft-hilt-sip-correction-503-01.txt

Here is a quick change log:
- Added a discussion on server maintenance vs. overload control
- Proposed an alternative change (alternative 1) that is based on 
introducing an new response code for overload.

All comments, thoughts are very welcome.

Thanks,

Volker




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From sip-bounces@ietf.org Tue Jul 10 17:14:12 2007
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Subject: Re: [Sip] Comments draft-ietf-sip-saml-02 - marc willekens
	-	"clarification" items
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Hi Jeff,
Hi Marc,

a comment below:

Jeff Hodges wrote:
> > -------- Original Message --------
> > Subject:     [Sip] Comments draft-ietf-sip-saml-02
> > Date:     Thu, 21 Jun 2007 16:47:46 +0200
> > From:     Willekens, Marc <marc.willekens@nsn.com>
> > To:     <sip@ietf.org>
> >
> >
> > [Page 3], Introduction
> >
> > Is this document only about assertion for the authorization or also for
> > the authentication?
>
> authorization.
>
>
> > [Page 12]
> >
> > Domain name for Alice: example.com or foo.com?? (same comment for other
> > figures)
>
> oops, good catch. foo.com.
>
>
> > Authentication service or Assertion service or both?
>
> authentication service -- the term is from rfc4474.
>
>
> > [Page 13], 6.1. Assertion Fetch profile
> >
> > Is this the only planned profile or will other profiles be defined and
> > added later?
>
> My personal present thinking is that there are other possible SIP SAML 
> profiles, e.g. see the appendix in -sip-saml-02 of other use cases, 
> but that they should likely be specified separately in their own drafts.
>
We have investigated other profiles. The SIP SAML profile has the very 
nice characteristic that it is nicely aligned with SIP Identity -- an 
already understood and agreed model.

> > Will there also be a push based model? Will it then be
> > "pushed" as part of the SIP message?
>
> This is something for the WG to think about. One _can_, in theory, 
> have the UAC assert some information and attach a SAML assertion to a 
> SIP request (e.g. Invite) body making such a statement. Thus this 
> approach could only be used by UACs because (1) only UACs may place 
> data in the SIP request body, and (2) there are vociferous objections 
> in the SIP WG to schemes whereby SIP intermediaries may place BLOBs 
> (binary large objects) in SIP header fields. See also..
>
> [Sip] wrt SIP Saml Review by Jiri Kuthan - "missing scenarios"
> http://www1.ietf.org/mail-archive/web/sip/current/msg19793.html
>
>
> We could specify this for UACs-only, and the other http-uri-based 
> fetch approach for, as it is currently, intermediates or UACs.
>
> However, before we go to the effort, what's the in-actuality demand 
> for doing so? Has anyone implemented the rfc4474 authentication 
> service role in the UAC and is it being used?
>
>

With regard to the ability of the SIP UA to attach an assertion there is 
another document that describes an extension to this SAML profile.
See http://tools.ietf.org/html/draft-jennings-sipping-pay-05

> > [Page 13], 6.1.1. URN
> >
> > What's the current consensus for the urn?
>
> No one has objected to what is in the spec, although no one has 
> commented specifically on it as yet, either.
>
>
>
> > [Page 17], 6.1.3.
> >
> > ...sender and Authentication service (AS) may be separate or... ???
> >
> > How will the AS and sender be combined?
>
> See the last paragraph on page 5 of rfc4474..
>
>   "This specification allows either a user agent or a proxy server to
>    provide identity services and to verify identities."
>
But for the use case where the end host attaches the assertion it might 
be more realistic to assume that the assertion is added to the body of 
the SIP message. This is currently not provided. This again is described 
in the SIP Pay draft.

>
> > [Page 22], 6.1.4.1.4.
> >
> > AttributeStatement is indicated. But what about Authentication and
> > Authorization Decision statement?
>
> This particular SIP SAML Profile doesn't specify use of those 
> statements, but they are not explicitly excluded either. Conceivably 
> someone could cook up a use case for them and make use of it in a 
> site-specific manner, or we might find use case and then can specify 
> their use more easily.
>
> If we do want to explicitly do this we should perhaps add that such 
> statements MAY be present, but their meaning is defined outside of 
> this profile.
>
>
We have received this question several times already. I believe we 
should explicitly enable the support since the AuthStatement provides 
more information than SIP Identity.


> > [Page31]
> >
> > This RFC draft talks about assertions. But how can an identity be
> > asserted only by considering a device, e.g. in case a device is stolen
> > or temporary used by someone else? At VON 2000, H. Schulzrinne has
> > already indicated that something as fingerprint has to be added for a
> > device with a changing set of owners.
>
> Although a valid thought, that being the question of the nature of the 
> actual authn mechanism between the UAC and the registration service, 
> it is out of scope of this specification. In fact, this draft nowhere 
> states that the subject's identity is asserted only by the device (UAC).
>
That's clearly outside the scope of this document.

>
> thanks for your review,
>
> =JeffH
>
Ciao
Hannes

>
>
>
>
>
>
>
>
> _______________________________________________
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From sip-bounces@ietf.org Tue Jul 10 17:44:24 2007
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The IESG has received a request from the Session Initiation Protocol WG 
(sip) to consider the following document:

- 'Requesting Answering Modes for the Session Initiation Protocol  
   (SIP) '
   <draft-ietf-sip-answermode-04.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to the
ietf@ietf.org mailing lists by 2007-08-05. Exceptionally, 
comments may be sent to iesg@ietf.org instead. In either case, please 
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-answermode-04.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=14099&rfc_flag=0



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From sip-bounces@ietf.org Tue Jul 10 21:24:42 2007
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Vijay K. Gurbani wrote:
> Dean Willis wrote:
> vkg> It looks like the drafts and a substantive body of the work in
> vkg> this area is out there; just needs to be organized for a more
> vkg> amenable deadline.  But, I may be mistaken...
>>
>> Otherwise said, you think the milestones in question were essentially
>> made redundandt by the adoption of the more-focused drafts? 
> 
> No; I thought that the focused drafts were *leading* to the milestone.
> But this may be my mis-interpretation of how a WG is chartered.  Is
> it the case that it is one draft per milestone?  Or could a series
> of drafts satisfy one milestone?

Depends on how your milestones are written -- ours are generally
structured "one milestone, one draft" rather like riots and Texas Rangers.

Usually a charter will have major planks, like "Clarify SIP security"
and then milestones (aka "deliverables") will be approved if they
support a plank.

> Doesn't S15 of sip-hitchhikers-guide-03
> (http://tools.ietf.org/html/draft-ietf-sip-hitchhikers-guide-03#section-15)
> come close to providing some structure to the various security-
> related drafts?  Plus, if sip-hitchhikers-guide is the last document
> that the WG produces, then it will keep up -- thanks to Jonathan --
> with newer technologies (SAML), protocols (DTLS) and documents as
> they are adopted by the WG.

Good question.

--
Dean



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From sip-bounces@ietf.org Tue Jul 10 21:32:08 2007
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Subject: [Sip] Need reviewers for draft-ietf-lemonade-streaming-02
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This is a pretty interesting draft, and I'd like to get a couple of
"sip perspective" reviewers to take a look at it and provide feedback to
the lemonade working group.

http://www.ietf.org/internet-drafts/draft-ietf-lemonade-streaming-02.txt



Abstract

   This document describes a method for streaming multimedia attachments
   received by a resource constrained and/or mobile device from an IMAP
   server.  It allows such clients, which often have limits in storage
   space and bandwidth, to play video and audio e-mail content.

   The document describes a profile for making use of the IMAP URLAUTH
   extension (RFC 4467), the Network Announcement SIP Media Service (RFC
   4240), and the Media Server Control Markup Language (RFC 4722).  The
   document also defines a new IMAP METADATA entry.


if you're willing to perform a review, please contact myself and Eric
Burger.

Thanks,

--
Dean


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From sip-bounces@ietf.org Wed Jul 11 02:20:48 2007
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Subject: [Sip] Need clarification on Retry-After header being sent as part
	of 500 response to second INVITE with lower Cseq number
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Hi,

   In RFC 3261, Section 14.2, Para 2, it's mentioned that Retry-After
header must be sent as part of 500 response to a second INVITE with
lower Cseq number.

   In the same RFC, section 20.1, Table 3, it's mentioned that sending
of Retry-After header is optional while sending a 500 response.

   I feel the statements are contradicting each other. Could you please clarify?

   If the first case is an exception for sending Retry-After header,
then the same should have been mentioned in Table 3 as well.

Thanks,
Jagan


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From sip-bounces@ietf.org Wed Jul 11 04:19:37 2007
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Subject: RE: [Sip] Last Call: draft-ietf-sip-answermode (Requesting Answering
	Modes for the Session Initiation Protocol (SIP)) to Proposed Standard
Date: Wed, 11 Jul 2007 01:18:44 -0700
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	Standard
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Section 4.3.2 states:

   The Answer-Mode and Priv-Answer-Mode header fields have equivalent
   functions, except that Priv-Answer-Mode requests a higher level of
   privilege in granting the answering mode specified by the request.

Would it not make sense to have a single header that performs both of
these equivalent functions, using a parameter (e.g. 'high' or
'level=3Dhigh') to distinguish between the two?

This might have the following benefits:

1) The document would be smaller.
2) The implementation may be simpler.
3) If the need for more than just two levels of privilege were found in
the future then this could be easily extended, rather than following the
current pattern of needing another header, e.g.
Really-High-Priv-Answer-Mode:
4) 'Priv' might be misinterpreted to mean 'private' rather than
'privilege' by the unfamiliar.



=20

> -----Original Message-----
> From: The IESG [mailto:iesg-secretary@ietf.org]=20
> Sent: 10 July 2007 22:44
> To: IETF-Announce
> Cc: sip@ietf.org
> Subject: [Sip] Last Call: draft-ietf-sip-answermode=20
> (Requesting Answering Modes for the Session Initiation=20
> Protocol (SIP)) to Proposed Standard
>=20
> The IESG has received a request from the Session Initiation=20
> Protocol WG
> (sip) to consider the following document:
>=20
> - 'Requesting Answering Modes for the Session Initiation Protocol =20
>    (SIP) '
>    <draft-ietf-sip-answermode-04.txt> as a Proposed Standard
>=20
> The IESG plans to make a decision in the next few weeks, and=20
> solicits final comments on this action.  Please send=20
> substantive comments to the ietf@ietf.org mailing lists by=20
> 2007-08-05. Exceptionally, comments may be sent to=20
> iesg@ietf.org instead. In either case, please retain the=20
> beginning of the Subject line to allow automated sorting.
>=20
> The file can be obtained via
> http://www.ietf.org/internet-drafts/draft-ietf-sip-answermode-04.txt
>=20
>=20
> IESG discussion can be tracked via
> https://datatracker.ietf.org/public/pidtracker.cgi?command=3Dvie
> w_id&dTag=3D14099&rfc_flag=3D0
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Wed Jul 11 05:03:07 2007
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Subject: [Sip] SIP SAML IANA Consideration Sction
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I compiled a first version of the IANA consideration section.
Please find it here: http://www.tschofenig.priv.at:8080/saml-sip/issue22

Ciao
Hannes



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From sip-bounces@ietf.org Wed Jul 11 07:25:57 2007
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Subject: AW: [Sip] Updated acr-code
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Dear all,
I still have a problem that the "header" privacy value is not included.
Due to the lack of a correct description what header privacy means =
within RFC3323 my understanding is still that header also includes all =
headers that should be deleted or anonymised when the "user" privacy =
values is used.

And it is also not clear if header includes the "id" privacy or not.

With regard to this arguments the ACR Response has also to be sent when =
"header" is used.

What to do now?

Also there is a different understanding what is meant by anonymous =
within the world. That is a regulation dependent issue. So From my =
understanding the use of the 433 should not to restrictive. If somebody =
want to use the header within his domain then let him use it.
That is my understanding.

So one solution could be that a sentence like based on network policy or =
regulatory requirements ACR could also apply based on other indications =
sent like the 'header' private value.

An other issue is that there should be work started to clarify the use =
of RFC 3323, 3325, 4244 and others defining privacy values.

Best Regards

Roland

Deutsche Telekom AG
T-Com, Zentrum Technik Einf=FChrung
Roland Jesske
Gateways, Protokolle, Konvergenz, TE33-6
Deutsche-Telekom-Allee 1, 64295 Darmstadt, Raum 314(Besucheradresse)
Postfach, 64307 Darmstadt (Postanschrift)
06151 83-5940 (Tel.)
06151 83-4577 (Fax)
E-Mail: r.jesske@t-com.net
http://www.t-com.de

Registerrechtliche Unternehmensangaben:
Deutsche Telekom AG
Aufsichtsrat: Dr. Klaus Zumwinkel (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),=20
Dr. Karl-Gerhard Eick (stellvertretender=20
Vorsitzender), Hamid Akhavan, Timotheus H=F6ttges, L
othar Pauly, Thomas Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,
Sitz der Gesellschaft: Bonn
WEEE-Reg.-Nr.: DE50478376
***************************************************************
Deutsche Telekom AG
Supervisory Board: Dr. Klaus Zumwinkel (Chairman)
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),
Hamid Akhavan, Timotheus H=F6ttges,=20
Lothar Pauly,  Thomas Sattelberger
Commercial register: Amtsgericht Bonn HRB 6794,
corporate seat: Bonn
WEEEreg.no: DE50478376


> -----Urspr=FCngliche Nachricht-----
> Von: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Gesendet: Freitag, 6. Juli 2007 04:16
> An: IETF SIP List
> Betreff: [Sip] Updated acr-code
>=20
>=20
> I've submitted an update to the acr-code draft. You can pick it up at:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-05.txt
>=20
> This draft addresses AD comments raised during AD evaluation. The=20
> primary change is that, previously, the specification=20
> indicated that a=20
> P-A-ID header field with an anonymous URI in it would count as an=20
> anonymous request. However, P-A-ID doesn't reflect what the=20
> user wants,=20
> which is strictly in the From. Consequently, PAID is really=20
> irrelevant=20
> here. So, this condition was removed.
>=20
> -Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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inline.

DRAGE, Keith (Keith) wrote:
> Two comments on the sigcomp part of this document.
> 
> While Sigcomp is mentioned in passing, RFC 3320 is referenced and this
> has been updated by RFC 4896. The ROHC people also consider RFC 4077 an
> integral part of the Sigcomp package and RFC 4896 assumes this.

I added a reference to 4896 along with 3320 as the reference for 
sigcomp. However our objective is not to do the hitchhikers guide to 
sigcomp; so as long as rfc4896 references 4077 people can follow that chain.

> 
> Additionally draft-ietf-rohc-sigcomp-sip is currently with IESG and
> this, among other things, defines a SIP Via header parameter and this
> should probably be referenced as a SIP extension. 

What???

Can this be???

Keith - did you let another working group define a SIP extension?

;)

Reference added.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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DRAGE, Keith (Keith) wrote:
> Your proposal as to what goes into the hitchhiker's guide sounds
> sensible to me.
> 
> I will take the comment about the file nameing convention into the
> next version of my essential-corrections draft. Proposals for a
> naming convention welcome; if no alternatives I will use something
> based on what Jonathan suggests below. I see only two issues:
> 
> -	we seem to be unable to decide what is a fix and what is not. If we
> make a decision that something is a fix after the -00 version, we
> need to decide whether the continuity of the existing nameing has
> priority or the convention of having -fix in the name.

I think the convention has priority. SO if we decide its not a fix, it 
gets renamed.

> 
> -	there are already a number of candidate drafts, which obviously do
> not yet follow this convention. Again we need to decide whether to
> keep the continuity of the existing naming or not.

I favor the convention.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Jesske, R wrote:
> Dear all, I still have a problem that the "header" privacy value is
> not included. Due to the lack of a correct description what header
> privacy means within RFC3323 my understanding is still that header
> also includes all headers that should be deleted or anonymised when
> the "user" privacy values is used.

My understanding is different.

> 
> And it is also not clear if header includes the "id" privacy or not.

Well, some recent work has attempted to clarify all of this:

http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-guideline-00.txt

though I disagree with parts of it, since there is really unclear 
relationship between these values. Its still not obvious what they mean. 
Indeed, according to the draft, 'header' does include P-A-ID, but 'user' 
does not. SO based on that I should include header in the list in 
acr-code. But I think it needs to be reversed.

> 
> With regard to this arguments the ACR Response has also to be sent
> when "header" is used.
> 
> What to do now?

Well, one answer is to punt this entirely. Acr-code can define a more 
abstract normative rule, and then privacy-guideline, should it be 
adopted, can have text that is very explicit on how it relates to ACR-code.


> 
> Also there is a different understanding what is meant by anonymous
> within the world. That is a regulation dependent issue. So From my
> understanding the use of the 433 should not to restrictive. If
> somebody want to use the header within his domain then let him use
> it. That is my understanding.
> 
> So one solution could be that a sentence like based on network policy
> or regulatory requirements ACR could also apply based on other
> indications sent like the 'header' private value.

The danger here is that we could end up with non-interoperable systems. 
If the UAC things a privacy value is NOT implying user anonymity, but a 
proxy in the terminating domain does think it means that, and thus it 
sends a 433, the UAC has no remedy and it thinks an error has occurred.

-JOnathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: RE: [Sip] I-D Action:draft-ietf-sip-hitchhikers-guide-03.txt
Date: Wed, 11 Jul 2007 15:32:26 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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Jonathan wrote:

> DRAGE, Keith (Keith) wrote:
> > Two comments on the sigcomp part of this document.
> >=20
> > While Sigcomp is mentioned in passing, RFC 3320 is=20
> referenced and this=20
> > has been updated by RFC 4896. The ROHC people also consider=20
> RFC 4077=20
> > an integral part of the Sigcomp package and RFC 4896 assumes this.
>=20
> I added a reference to 4896 along with 3320 as the reference=20
> for sigcomp. However our objective is not to do the=20
> hitchhikers guide to sigcomp; so as long as rfc4896=20
> references 4077 people can follow that chain.
>=20
Fine.

> > Additionally draft-ietf-rohc-sigcomp-sip is currently with IESG and=20
> > this, among other things, defines a SIP Via header=20
> parameter and this=20
> > should probably be referenced as a SIP extension.
>=20
> What???
>=20
> Can this be???
>=20
> Keith - did you let another working group define a SIP extension?
>=20
> ;)
> =20
It followed the IANA registry rules for SIP header parameters, which do =
not require it to come to SIP. However in parallel with the ROHC WGLC, =
we did issue the WGLC on the SIP list, as we did ask Adam Roach to do a =
review as a designated SIP expert. If people want to handle these =
differently in the future them please shout.

> Reference added.

Fine

Regards

Keith

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Wednesday, July 11, 2007 2:19 PM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D Action:draft-ietf-sip-hitchhikers-guide-03.txt
>=20
> inline.
>=20
> DRAGE, Keith (Keith) wrote:
> > Two comments on the sigcomp part of this document.
> >=20
> > While Sigcomp is mentioned in passing, RFC 3320 is=20
> referenced and this=20
> > has been updated by RFC 4896. The ROHC people also consider=20
> RFC 4077=20
> > an integral part of the Sigcomp package and RFC 4896 assumes this.
>=20
> I added a reference to 4896 along with 3320 as the reference=20
> for sigcomp. However our objective is not to do the=20
> hitchhikers guide to sigcomp; so as long as rfc4896=20
> references 4077 people can follow that chain.
>=20
> >=20
> > Additionally draft-ietf-rohc-sigcomp-sip is currently with IESG and=20
> > this, among other things, defines a SIP Via header=20
> parameter and this=20
> > should probably be referenced as a SIP extension.
>=20
> What???
>=20
> Can this be???
>=20
> Keith - did you let another working group define a SIP extension?
>=20
> ;)
>=20
> Reference added.
>=20
> -Jonathan R.
>=20
>=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



From sip-bounces@ietf.org Wed Jul 11 09:44:29 2007
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Subject: RE: [Sip] Updated acr-code
Date: Wed, 11 Jul 2007 15:43:14 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Jesske, R" <R.Jesske@t-com.net>, <sip@ietf.org>
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Please note that this document is formally in IESG last call. Therefore =
any comments that you want to "stick" should also be made to =
ietf@ietf.org in accordance with the following instructions. That =
doesn't preclude discussion on this list - just means that they may not =
be taken into account.

Regards

Keith

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

- 'Rejecting Anonymous Requests in the Session Initiation Protocol (SIP) =
'
   <draft-ietf-sip-acr-code-05.txt> as a Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits =
final comments on this action.  Please send substantive comments to the =
ietf@ietf.org mailing lists by 2007-07-19. Exceptionally, comments may =
be sent to iesg@ietf.org instead. In either case, please retain the =
beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-05.txt


IESG discussion can be tracked via
https://datatracker.ietf.org/public/pidtracker.cgi?command=3Dview_id&dTag=
=3D14158&rfc_flag=3D0=20

> -----Original Message-----
> From: Jesske, R [mailto:R.Jesske@t-com.net]=20
> Sent: Wednesday, July 11, 2007 12:25 PM
> To: jdrosen@cisco.com; sip@ietf.org
> Subject: AW: [Sip] Updated acr-code
>=20
> Dear all,
> I still have a problem that the "header" privacy value is not=20
> included.
> Due to the lack of a correct description what header privacy=20
> means within RFC3323 my understanding is still that header=20
> also includes all headers that should be deleted or=20
> anonymised when the "user" privacy values is used.
>=20
> And it is also not clear if header includes the "id" privacy or not.
>=20
> With regard to this arguments the ACR Response has also to be=20
> sent when "header" is used.
>=20
> What to do now?
>=20
> Also there is a different understanding what is meant by=20
> anonymous within the world. That is a regulation dependent=20
> issue. So From my understanding the use of the 433 should not=20
> to restrictive. If somebody want to use the header within his=20
> domain then let him use it.
> That is my understanding.
>=20
> So one solution could be that a sentence like based on=20
> network policy or regulatory requirements ACR could also=20
> apply based on other indications sent like the 'header' private value.
>=20
> An other issue is that there should be work started to=20
> clarify the use of RFC 3323, 3325, 4244 and others defining=20
> privacy values.
>=20
> Best Regards
>=20
> Roland
>=20
> Deutsche Telekom AG
> T-Com, Zentrum Technik Einf=FChrung
> Roland Jesske
> Gateways, Protokolle, Konvergenz, TE33-6=20
> Deutsche-Telekom-Allee 1, 64295 Darmstadt, Raum=20
> 314(Besucheradresse) Postfach, 64307 Darmstadt (Postanschrift)
> 06151 83-5940 (Tel.)
> 06151 83-4577 (Fax)
> E-Mail: r.jesske@t-com.net
> http://www.t-com.de
>=20
> Registerrechtliche Unternehmensangaben:
> Deutsche Telekom AG
> Aufsichtsrat: Dr. Klaus Zumwinkel (Vorsitzender)
> Vorstand: Ren=E9 Obermann (Vorsitzender), Dr. Karl-Gerhard Eick=20
> (stellvertretender Vorsitzender), Hamid Akhavan, Timotheus=20
> H=F6ttges, L othar Pauly, Thomas Sattelberger
> Handelsregister: Amtsgericht Bonn HRB 6794, Sitz der=20
> Gesellschaft: Bonn
> WEEE-Reg.-Nr.: DE50478376
> ***************************************************************
> Deutsche Telekom AG
> Supervisory Board: Dr. Klaus Zumwinkel (Chairman) Board of=20
> Management: Ren=E9 Obermann (Chairman), Dr. Karl-Gerhard Eick=20
> (Deputy Chairman), Hamid Akhavan, Timotheus H=F6ttges, Lothar=20
> Pauly,  Thomas Sattelberger Commercial register: Amtsgericht=20
> Bonn HRB 6794, corporate seat: Bonn
> WEEEreg.no: DE50478376
>=20
>=20
> > -----Urspr=FCngliche Nachricht-----
> > Von: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > Gesendet: Freitag, 6. Juli 2007 04:16
> > An: IETF SIP List
> > Betreff: [Sip] Updated acr-code
> >=20
> >=20
> > I've submitted an update to the acr-code draft. You can=20
> pick it up at:
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-acr-code-05.txt
> >=20
> > This draft addresses AD comments raised during AD evaluation. The=20
> > primary change is that, previously, the specification=20
> indicated that a=20
> > P-A-ID header field with an anonymous URI in it would count as an=20
> > anonymous request. However, P-A-ID doesn't reflect what the user=20
> > wants, which is strictly in the From. Consequently, PAID is really=20
> > irrelevant here. So, this condition was removed.
> >=20
> > -Jonathan R.
> > --=20
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Cisco Fellow                                   Parsippany, NJ=20
> > 07054-2711
> > Cisco Systems
> > jdrosen@cisco.com                              FAX:   (973) 952-5050
> > http://www.jdrosen.net                         PHONE: (973) 952-5000
> > http://www.cisco.com
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



From sip-bounces@ietf.org Wed Jul 11 10:09:44 2007
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Thread-Topic: Hop limit diagnostics
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(As WG chair)

We have a couple of related milestones on our charter that we are stuck
on:

Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS) =20
Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)=20

The draft associated with this expired some way back, but you can find
it at:

http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03

The charter item is for a more general document that covers other error
situations as well as hop limit issues.

However the editor's hit the intractable problem in that any transport
decision is made on the request on any particular hop, and if UDP is
used on the request, it will also be used on the response on any
particular hop. This was specified based on the assumption that any
response would not be significantly larger than the request, but as soon
as we start putting lots of useful diagnostic information in the
response, this no longer applies.

So we are now looking for the way forward. Options include:

A)	It is not worth the extra cycles - delete the milestone.

B)	Limit the diagnostic information (to say around 100 bytes in the
worst case). If so will it contain enough useful information to make it
usable.

C)	Solve the transport problem. And no, we do not have a debate
here on deprecating UDP. We've been there and done that.

Unless people can come up with something that looks achievable, the
working group chairs are currently favouring A) above.

Comments please.


Keith


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This list is for NEW development of the core SIP Protocol
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Subject: Re: [Sip] Proposed: delete Guidelines for use of SIP
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On Tue, 2007-07-10 at 20:28 -0500, Dean Willis wrote:
> Vijay K. Gurbani wrote:
> > Dean Willis wrote:
> > vkg> It looks like the drafts and a substantive body of the work in
> > vkg> this area is out there; just needs to be organized for a more
> > vkg> amenable deadline.  But, I may be mistaken...
> >>
> >> Otherwise said, you think the milestones in question were essentially
> >> made redundandt by the adoption of the more-focused drafts? 
> > 
> > No; I thought that the focused drafts were *leading* to the milestone.
> > But this may be my mis-interpretation of how a WG is chartered.  Is
> > it the case that it is one draft per milestone?  Or could a series
> > of drafts satisfy one milestone?
> 
> Depends on how your milestones are written -- ours are generally
> structured "one milestone, one draft" rather like riots and Texas Rangers.

Well, I think that the drafts Vijay listed are all parts of that
picture.  Perhaps what is needed is to get a volunteer to edit a
document that pulls them all together into a single document to address
the deliverable?

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXecs project coordinator - SIPfoundry http://www.sipfoundry.org/sipXecs
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/



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From sip-bounces@ietf.org Wed Jul 11 11:07:09 2007
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Subject: Re: [Sip] Hop limit diagnostics
Date: Wed, 11 Jul 2007 10:06:42 -0500
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For completeness - haven't we also talked about requests in the  
opposite direction and diagnostics
by indirection? I know each of those have their own thorny problems,  
but the tradeoff against really
large responses may make them the least unpalatable.

And I remember conversations, but not much list traffic about a  
series of best-effort provisionals
to carry the information.

I still think its worth figuring out, but won't throw fits if the WG  
decides to put it off to the indefinite future.

RjS


On Jul 11, 2007, at 9:07 AM, DRAGE, Keith ((Keith)) wrote:

> (As WG chair)
>
> We have a couple of related milestones on our charter that we are  
> stuck
> on:
>
> Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS)
> Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)
>
> The draft associated with this expired some way back, but you can find
> it at:
>
> http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03
>
> The charter item is for a more general document that covers other  
> error
> situations as well as hop limit issues.
>
> However the editor's hit the intractable problem in that any transport
> decision is made on the request on any particular hop, and if UDP is
> used on the request, it will also be used on the response on any
> particular hop. This was specified based on the assumption that any
> response would not be significantly larger than the request, but as  
> soon
> as we start putting lots of useful diagnostic information in the
> response, this no longer applies.
>
> So we are now looking for the way forward. Options include:
>
> A)	It is not worth the extra cycles - delete the milestone.
>
> B)	Limit the diagnostic information (to say around 100 bytes in the
> worst case). If so will it contain enough useful information to  
> make it
> usable.
>
> C)	Solve the transport problem. And no, we do not have a debate
> here on deprecating UDP. We've been there and done that.
>
> Unless people can come up with something that looks achievable, the
> working group chairs are currently favouring A) above.
>
> Comments please.
>
>
> Keith
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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A New Internet-Draft is available from the on-line Internet-Drafts 
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This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Location Conveyance for the Session Initiation Protocol
	Author(s)	: J. Polk, B. Rosen
	Filename	: draft-ietf-sip-location-conveyance-08.txt
	Pages		: 38
	Date		: 2007-7-11
	
This document defines an extension to the Session Initiation 
   Protocol (SIP) to convey geographic location information from one 
   SIP entity to another SIP entity.  The extension covers end to end 
   conveyance as well as location-based routing, where proxy servers 
   make routing decisions based on the location of the SIP user agents.

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	Title		: Certificate Management Service for The Session Initiation Protocol (SIP)
	Author(s)	: C. Jennings, et al.
	Filename	: draft-ietf-sip-certs-04.txt
	Pages		: 30
	Date		: 2007-7-11
	
This draft defines a Credential Service that allows Session
   Initiation Protocol (SIP) User Agents (UAs) to use a SIP package to
   discover the certificates of other users.  This mechanism allows user
   agents that want to contact a given Address-of-Record (AOR) to
   retrieve that AOR's certificate by subscribing to the Credential
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   certificate.  The Credential Service also allows users to store and
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	Title		: End-to-middle Security in the Session Initiation Protocol (SIP)
	Author(s)	: K. Ono, S. Tachimoto
	Filename	: draft-ietf-sip-e2m-sec-06.txt
	Pages		: 30
	Date		: 2007-7-11
	
Some services provided by intermediaries depend on their ability to
   inspect a message body in the Session Initiation Protocol (SIP).
   When sensitive information is included in the message body, a SIP
   User Agent (UA) needs to protect it from other intermediaries than
   those that the UA agreed to disclose it to.  This document proposes a
   mechanism for securing information passed between an end user and
   intermediaries using S/MIME.  It also proposes mechanisms for a UA to
   discover intermediaries which need to inspect an S/MIME-secured
   message body, or to receive the message body with data integrity

   This specification is approved at the proposed standards level due to
   the IANA registration requirements.  Is is of sufficient quality for
   that level, however, the use of this mechanism in this specification
   are considered experimental.

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_______________________________________________
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From sip-bounces@ietf.org Wed Jul 11 14:33:03 2007
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From: Sean Olson <Sean.Olson@microsoft.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "sip@ietf.org"
	<sip@ietf.org>
Date: Wed, 11 Jul 2007 11:32:39 -0700
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Why not:

D) Define a diagnostic information mechanism that works with TCP and accept=
 that it will not work with UDP



-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
Sent: Wednesday, July 11, 2007 7:07 AM
To: sip@ietf.org
Subject: [Sip] Hop limit diagnostics

(As WG chair)

We have a couple of related milestones on our charter that we are stuck
on:

Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS)
Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)

The draft associated with this expired some way back, but you can find
it at:

http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03

The charter item is for a more general document that covers other error
situations as well as hop limit issues.

However the editor's hit the intractable problem in that any transport
decision is made on the request on any particular hop, and if UDP is
used on the request, it will also be used on the response on any
particular hop. This was specified based on the assumption that any
response would not be significantly larger than the request, but as soon
as we start putting lots of useful diagnostic information in the
response, this no longer applies.

So we are now looking for the way forward. Options include:

A)      It is not worth the extra cycles - delete the milestone.

B)      Limit the diagnostic information (to say around 100 bytes in the
worst case). If so will it contain enough useful information to make it
usable.

C)      Solve the transport problem. And no, we do not have a debate
here on deprecating UDP. We've been there and done that.

Unless people can come up with something that looks achievable, the
working group chairs are currently favouring A) above.

Comments please.


Keith


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From sip-bounces@ietf.org Wed Jul 11 15:42:59 2007
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Subject: RE: [Sip] Re: Conveyance -08 and Geolocation-Error header
Date: Wed, 11 Jul 2007 21:42:13 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
	"Paul Kyzivat" <pkyzivat@cisco.com>
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You wrote:

> In general, I think we have learned that we should help=20
> people with debugging distributed systems with distributed=20
> responsibility, as otherwise they'll either struggle or=20
> invent private mechanisms to include this information. (See=20
> SER for the the latter.)=20

If we are not careful here we end up back in the issue of responses
being significantly larger than the request sent on a particular hop,
and therefore the old problem of what happens on UDP when this occurs.

Regards

Keith

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]=20
> Sent: Saturday, July 07, 2007 1:06 PM
> To: Paul Kyzivat
> Cc: sip@ietf.org; James M. Polk
> Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
>=20
>=20
> On Jul 6, 2007, at 11:24 PM, Paul Kyzivat wrote:
>=20
> > Is it really reasonable to expect a proxy that inserts a=20
> location to=20
> > remember what location it inserted, in enough detail to=20
> diagnose why=20
> > it might have been messed up? I presume it would have to=20
> reconstruct=20
> > what the location would have been based in information in the=20
> > response. It may not be possible to reconstruct the exact=20
> same value,=20
> > because the UAC may have moved in the meantime. I guess the=20
> inserted=20
> > location would have to be by reference, but the URL of the=20
> reference=20
> > may itself not be easily reconstructed, and multiple=20
> queries of it may=20
> > not return the same result.
> >
> > IMO it would be much clearer to include the location itself in the=20
> > response. And if the location had been passed by reference,=20
> it might=20
> > be better to pass it by value in the response, so that there is no=20
> > possibility of the value changing.
> >
>=20
> I tend to agree that this would be better. The=20
> reference-to-value translation may not always be feasible: If=20
> a proxy inserts a location reference and a downstream proxy=20
> finds a problem, it can't insert a value (except by the data=20
> URL mechanism I have been mentioning...).
>=20
>=20
> > OTOH that might result in divulging the location to nodes=20
> upstream of=20
> > the one that included it. If it was passed by reference=20
> then the node=20
> > that had inserted it could remove the reference in the=20
> response. But=20
> > if it is passed back by value that isn't allowed.
> >
> > The alternative is that the proxy will have to retain the location=20
> > until the completion of the transaction. Maybe that's ok,=20
> but if its=20
> > only to diagnose an occasional glitch I get it doesn't get done.
>=20
> My assumption was that most locations will be inserted by the=20
> UAC rather than a proxy. In those cases, retaining the=20
> information is probably not a big deal, since the UAC will=20
> generally only have one.
>=20
> In summary, I think returning as much information about the=20
> erroneous location information as possible seems best, i.e.,=20
> both the location information and the party that inserted it.=20
> This avoids the "who the heck put that bogus location=20
> information in there?" problem.
>=20
> In general, I think we have learned that we should help=20
> people with debugging distributed systems with distributed=20
> responsibility, as otherwise they'll either struggle or=20
> invent private mechanisms to include this information. (See=20
> SER for the the latter.)
>=20
>=20
> >
> > 	Paul
> >
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sip] Re: Conveyance -08 and Geolocation-Error header
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At 02:42 PM 7/11/2007, DRAGE, Keith \(Keith\) wrote:
>You wrote:
>
> > In general, I think we have learned that we should help
> > people with debugging distributed systems with distributed
> > responsibility, as otherwise they'll either struggle or
> > invent private mechanisms to include this information. (See
> > SER for the the latter.)
>
>If we are not careful here we end up back in the issue of responses
>being significantly larger than the request sent on a particular hop,
>and therefore the old problem of what happens on UDP when this occurs.

one crazy (or stupid?) idea is

- now that there is a more robust Geolocation-Error header in 
responses (verses a Reason or Warning header), a *(SEMI CAType) could 
be added to it to allow a Location Recipient to include each CAType 
it had a problem with or wasn't there that was considered necessary 
to use the location in the first place.  This idea doesn't include 
the element values, just the elements - which saves bytes, and 
doesn't reveal the element values either.  It also doesn't include a 
message body in the response.

I think this is more information than is presently there, making for 
better decisions by location inserters or for troubleshooting.  The 
only security risk I see with this is the listing of some of the 
CAtypes that were in the original request.  Presumably there were 
other elements in the request, so little (if anything) can be 
extracted from this knowledge if intercepted.


>Regards
>
>Keith
>
> > -----Original Message-----
> > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> > Sent: Saturday, July 07, 2007 1:06 PM
> > To: Paul Kyzivat
> > Cc: sip@ietf.org; James M. Polk
> > Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
> >
> >
> > On Jul 6, 2007, at 11:24 PM, Paul Kyzivat wrote:
> >
> > > Is it really reasonable to expect a proxy that inserts a
> > location to
> > > remember what location it inserted, in enough detail to
> > diagnose why
> > > it might have been messed up? I presume it would have to
> > reconstruct
> > > what the location would have been based in information in the
> > > response. It may not be possible to reconstruct the exact
> > same value,
> > > because the UAC may have moved in the meantime. I guess the
> > inserted
> > > location would have to be by reference, but the URL of the
> > reference
> > > may itself not be easily reconstructed, and multiple
> > queries of it may
> > > not return the same result.
> > >
> > > IMO it would be much clearer to include the location itself in the
> > > response. And if the location had been passed by reference,
> > it might
> > > be better to pass it by value in the response, so that there is no
> > > possibility of the value changing.
> > >
> >
> > I tend to agree that this would be better. The
> > reference-to-value translation may not always be feasible: If
> > a proxy inserts a location reference and a downstream proxy
> > finds a problem, it can't insert a value (except by the data
> > URL mechanism I have been mentioning...).
> >
> >
> > > OTOH that might result in divulging the location to nodes
> > upstream of
> > > the one that included it. If it was passed by reference
> > then the node
> > > that had inserted it could remove the reference in the
> > response. But
> > > if it is passed back by value that isn't allowed.
> > >
> > > The alternative is that the proxy will have to retain the location
> > > until the completion of the transaction. Maybe that's ok,
> > but if its
> > > only to diagnose an occasional glitch I get it doesn't get done.
> >
> > My assumption was that most locations will be inserted by the
> > UAC rather than a proxy. In those cases, retaining the
> > information is probably not a big deal, since the UAC will
> > generally only have one.
> >
> > In summary, I think returning as much information about the
> > erroneous location information as possible seems best, i.e.,
> > both the location information and the party that inserted it.
> > This avoids the "who the heck put that bogus location
> > information in there?" problem.
> >
> > In general, I think we have learned that we should help
> > people with debugging distributed systems with distributed
> > responsibility, as otherwise they'll either struggle or
> > invent private mechanisms to include this information. (See
> > SER for the the latter.)
> >
> >
> > >
> > >     Paul
> > >
> >
> >
> >
> > _______________________________________________
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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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From sip-bounces@ietf.org Wed Jul 11 17:58:13 2007
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	Modes for the Session Initiation Protocol (SIP)) to Proposed Standard
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Thanks for the feedback.

Steve Langstaff wrote:
> Section 4.3.2 states:
> 
>    The Answer-Mode and Priv-Answer-Mode header fields have equivalent
>    functions, except that Priv-Answer-Mode requests a higher level of
>    privilege in granting the answering mode specified by the request.
> 
> Would it not make sense to have a single header that performs both of
> these equivalent functions, using a parameter (e.g. 'high' or
> 'level=high') to distinguish between the two?

Sure would. If I recall correctly, that approach was taken by an earlier 
version of the draft, about ten revisions back.

For some reason I never fully understood, people argued that distinct 
header field names made for less of a security issue than having 
parameterized privilege levels. Since we don't have separate option tags 
for negotiating support for the header fields independently, it strikes 
me as making no difference either way.

Argument went back and forth for a while, and the WG seemed to develop 
consensus around the approach as documented (different header field names)

> This might have the following benefits:
> 
> 1) The document would be smaller.

True.

> 2) The implementation may be simpler.

Probably not.

> 3) If the need for more than just two levels of privilege were found in
> the future then this could be easily extended, rather than following the
> current pattern of needing another header, e.g.
> Really-High-Priv-Answer-Mode:

Also true, but the effort of documenting another header field name is no 
higher than that required to document a new parameter value on a header 
field name.

> 4) 'Priv' might be misinterpreted to mean 'private' rather than
> 'privilege' by the unfamiliar.

Also true, but it's shorter. The WG settled on this syntax as a 
compromise between readability and compactness.
--
dean


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From sip-bounces@ietf.org Wed Jul 11 20:33:43 2007
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
Date: Wed, 11 Jul 2007 20:33:17 -0400
To: "James M. Polk" <jmpolk@cisco.com>
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This could be done, but requires a mapping from XML tags to CAtypes  
at the proxy, which seems somewhat unnatural. Generally speaking, it  
seems likely that only one or two tags contradict each other or are  
"wrong", so that the saving is likely to be rather modest,  
particularly given that many of the civic XML tags are only two or  
three characters long. Thus, you might realize a saving of one or two  
bytes in most cases.

My major concern is identifying the source of insertion of the LI.  
This would be a domain name, typically 50 bytes or so, so I don't  
think this is a major issue.

On Jul 11, 2007, at 5:39 PM, James M. Polk wrote:

> At 02:42 PM 7/11/2007, DRAGE, Keith \(Keith\) wrote:
>> You wrote:
>>
>> > In general, I think we have learned that we should help
>> > people with debugging distributed systems with distributed
>> > responsibility, as otherwise they'll either struggle or
>> > invent private mechanisms to include this information. (See
>> > SER for the the latter.)
>>
>> If we are not careful here we end up back in the issue of responses
>> being significantly larger than the request sent on a particular hop,
>> and therefore the old problem of what happens on UDP when this  
>> occurs.
>
> one crazy (or stupid?) idea is
>
> - now that there is a more robust Geolocation-Error header in  
> responses (verses a Reason or Warning header), a *(SEMI CAType)  
> could be added to it to allow a Location Recipient to include each  
> CAType it had a problem with or wasn't there that was considered  
> necessary to use the location in the first place.  This idea  
> doesn't include the element values, just the elements - which saves  
> bytes, and doesn't reveal the element values either.  It also  
> doesn't include a message body in the response.
>
> I think this is more information than is presently there, making  
> for better decisions by location inserters or for troubleshooting.   
> The only security risk I see with this is the listing of some of  
> the CAtypes that were in the original request.  Presumably there  
> were other elements in the request, so little (if anything) can be  
> extracted from this knowledge if intercepted.
>
>
>> Regards
>>
>> Keith
>>
>> > -----Original Message-----
>> > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>> > Sent: Saturday, July 07, 2007 1:06 PM
>> > To: Paul Kyzivat
>> > Cc: sip@ietf.org; James M. Polk
>> > Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
>> >
>> >
>> > On Jul 6, 2007, at 11:24 PM, Paul Kyzivat wrote:
>> >
>> > > Is it really reasonable to expect a proxy that inserts a
>> > location to
>> > > remember what location it inserted, in enough detail to
>> > diagnose why
>> > > it might have been messed up? I presume it would have to
>> > reconstruct
>> > > what the location would have been based in information in the
>> > > response. It may not be possible to reconstruct the exact
>> > same value,
>> > > because the UAC may have moved in the meantime. I guess the
>> > inserted
>> > > location would have to be by reference, but the URL of the
>> > reference
>> > > may itself not be easily reconstructed, and multiple
>> > queries of it may
>> > > not return the same result.
>> > >
>> > > IMO it would be much clearer to include the location itself in  
>> the
>> > > response. And if the location had been passed by reference,
>> > it might
>> > > be better to pass it by value in the response, so that there  
>> is no
>> > > possibility of the value changing.
>> > >
>> >
>> > I tend to agree that this would be better. The
>> > reference-to-value translation may not always be feasible: If
>> > a proxy inserts a location reference and a downstream proxy
>> > finds a problem, it can't insert a value (except by the data
>> > URL mechanism I have been mentioning...).
>> >
>> >
>> > > OTOH that might result in divulging the location to nodes
>> > upstream of
>> > > the one that included it. If it was passed by reference
>> > then the node
>> > > that had inserted it could remove the reference in the
>> > response. But
>> > > if it is passed back by value that isn't allowed.
>> > >
>> > > The alternative is that the proxy will have to retain the  
>> location
>> > > until the completion of the transaction. Maybe that's ok,
>> > but if its
>> > > only to diagnose an occasional glitch I get it doesn't get done.
>> >
>> > My assumption was that most locations will be inserted by the
>> > UAC rather than a proxy. In those cases, retaining the
>> > information is probably not a big deal, since the UAC will
>> > generally only have one.
>> >
>> > In summary, I think returning as much information about the
>> > erroneous location information as possible seems best, i.e.,
>> > both the location information and the party that inserted it.
>> > This avoids the "who the heck put that bogus location
>> > information in there?" problem.
>> >
>> > In general, I think we have learned that we should help
>> > people with debugging distributed systems with distributed
>> > responsibility, as otherwise they'll either struggle or
>> > invent private mechanisms to include this information. (See
>> > SER for the the latter.)
>> >
>> >
>> > >
>> > >     Paul
>> > >
>> >
>> >
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol Use
>> > sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>> >



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At 07:33 PM 7/11/2007, Henning Schulzrinne wrote:
>This could be done, but requires a mapping from XML tags to CAtypes
>at the proxy, which seems somewhat unnatural. Generally speaking, it
>seems likely that only one or two tags contradict each other or are
>"wrong", so that the saving is likely to be rather modest,
>particularly given that many of the civic XML tags are only two or
>three characters long. Thus, you might realize a saving of one or two
>bytes in most cases.

but I was comparing the savings of having only one or two (as you 
said) "CAType=<blah>" added to the Geolocation-Error header verses 
having a full PIDF-LO added to a message body in the response. I 
think you'll agree that the size/bytes savings in comparison is 
substantial - and goes towards Keith's point of UDP.

This also is consistent with most folks thinking that location 
shouldn't be in a response.  I think this proposal can get us a long 
way towards what you want, without saying "you said you were at N 
Main St., which doesn't exist, instead of NW Main St, which does". 
The CAType for prefix street name would be what's included in the 
response, because it was the part of the civic location that was bad.


>My major concern is identifying the source of insertion of the LI.
>This would be a domain name, typically 50 bytes or so, so I don't
>think this is a major issue.
>
>On Jul 11, 2007, at 5:39 PM, James M. Polk wrote:
>
>>At 02:42 PM 7/11/2007, DRAGE, Keith \(Keith\) wrote:
>>>You wrote:
>>>
>>> > In general, I think we have learned that we should help
>>> > people with debugging distributed systems with distributed
>>> > responsibility, as otherwise they'll either struggle or
>>> > invent private mechanisms to include this information. (See
>>> > SER for the the latter.)
>>>
>>>If we are not careful here we end up back in the issue of responses
>>>being significantly larger than the request sent on a particular hop,
>>>and therefore the old problem of what happens on UDP when this
>>>occurs.
>>
>>one crazy (or stupid?) idea is
>>
>>- now that there is a more robust Geolocation-Error header in
>>responses (verses a Reason or Warning header), a *(SEMI CAType)
>>could be added to it to allow a Location Recipient to include each
>>CAType it had a problem with or wasn't there that was considered
>>necessary to use the location in the first place.  This idea
>>doesn't include the element values, just the elements - which saves
>>bytes, and doesn't reveal the element values either.  It also
>>doesn't include a message body in the response.
>>
>>I think this is more information than is presently there, making
>>for better decisions by location inserters or for troubleshooting.
>>The only security risk I see with this is the listing of some of
>>the CAtypes that were in the original request.  Presumably there
>>were other elements in the request, so little (if anything) can be
>>extracted from this knowledge if intercepted.
>>
>>
>>>Regards
>>>
>>>Keith
>>>
>>> > -----Original Message-----
>>> > From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>>> > Sent: Saturday, July 07, 2007 1:06 PM
>>> > To: Paul Kyzivat
>>> > Cc: sip@ietf.org; James M. Polk
>>> > Subject: Re: [Sip] Re: Conveyance -08 and Geolocation-Error header
>>> >
>>> >
>>> > On Jul 6, 2007, at 11:24 PM, Paul Kyzivat wrote:
>>> >
>>> > > Is it really reasonable to expect a proxy that inserts a
>>> > location to
>>> > > remember what location it inserted, in enough detail to
>>> > diagnose why
>>> > > it might have been messed up? I presume it would have to
>>> > reconstruct
>>> > > what the location would have been based in information in the
>>> > > response. It may not be possible to reconstruct the exact
>>> > same value,
>>> > > because the UAC may have moved in the meantime. I guess the
>>> > inserted
>>> > > location would have to be by reference, but the URL of the
>>> > reference
>>> > > may itself not be easily reconstructed, and multiple
>>> > queries of it may
>>> > > not return the same result.
>>> > >
>>> > > IMO it would be much clearer to include the location itself in
>>>the
>>> > > response. And if the location had been passed by reference,
>>> > it might
>>> > > be better to pass it by value in the response, so that there
>>>is no
>>> > > possibility of the value changing.
>>> > >
>>> >
>>> > I tend to agree that this would be better. The
>>> > reference-to-value translation may not always be feasible: If
>>> > a proxy inserts a location reference and a downstream proxy
>>> > finds a problem, it can't insert a value (except by the data
>>> > URL mechanism I have been mentioning...).
>>> >
>>> >
>>> > > OTOH that might result in divulging the location to nodes
>>> > upstream of
>>> > > the one that included it. If it was passed by reference
>>> > then the node
>>> > > that had inserted it could remove the reference in the
>>> > response. But
>>> > > if it is passed back by value that isn't allowed.
>>> > >
>>> > > The alternative is that the proxy will have to retain the
>>>location
>>> > > until the completion of the transaction. Maybe that's ok,
>>> > but if its
>>> > > only to diagnose an occasional glitch I get it doesn't get done.
>>> >
>>> > My assumption was that most locations will be inserted by the
>>> > UAC rather than a proxy. In those cases, retaining the
>>> > information is probably not a big deal, since the UAC will
>>> > generally only have one.
>>> >
>>> > In summary, I think returning as much information about the
>>> > erroneous location information as possible seems best, i.e.,
>>> > both the location information and the party that inserted it.
>>> > This avoids the "who the heck put that bogus location
>>> > information in there?" problem.
>>> >
>>> > In general, I think we have learned that we should help
>>> > people with debugging distributed systems with distributed
>>> > responsibility, as otherwise they'll either struggle or
>>> > invent private mechanisms to include this information. (See
>>> > SER for the the latter.)
>>> >
>>> >
>>> > >
>>> > >     Paul
>>> > >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> > This list is for NEW development of the core SIP Protocol Use
>>> > sip-implementors@cs.columbia.edu for questions on current sip
>>> > Use sipping@ietf.org for new developments on the application of sip
>>> >


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From sip-bounces@ietf.org Thu Jul 12 03:31:33 2007
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Subject: RE: [Sip] SIP Identity using Media Path
Date: Thu, 12 Jul 2007 08:30:56 +0100
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From: "Gilad Shaham" <gshaham@juniper.net>
To: "Dan Wing" <dwing@cisco.com>,
	"Hans Persson" <hasse@ingate.com>
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Hi,

Dan Wing Wrote:
> > > > Section 4.1, item about Contact:
> > > >
> > > > How can the Contact header usefully be used in the
> > signing process? An
> > > > SBC along the message path will happily replace it.
> > >
> > > I have removed that in -01 (which isn't yet published, of
> > course).  There
> > > was some thought it was necessary, but I agree it should be
removed
> > > from the signature.
> > >
> > > On a similar note, I am considering removing CallId from
> > the signature.
> > > Oftentimes the Call Id value contains an IP address (in
> > dotted decimal
> > > or hex), and an SBC or B2BUA may also want to rewrite such
> > a CallId.  I
> > > have made a note of that in -01 so this can be discussed.
> >
> > I thought about that as well, but apparently I forgot about
> > adding it to
> > my mail. I think it would be a good idea to take Call-ID out as
well.
>=20
> Ok, thanks.
>=20

I agree with this, but keep in mind that the call-id serves as a nonce
in RFC 4474. It might require introducing a new header/parameter to be
the nonce.

Regards,
Gilad


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From sip-bounces@ietf.org Thu Jul 12 05:26:29 2007
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Subject: RE: [Sip] Hop limit diagnostics
Date: Thu, 12 Jul 2007 11:23:51 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Robert Sparks" <rjsparks@nostrum.com>
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I believe content-indirection was looked at and also sending a new
request in the reverse direction.

I believe the issue with content indirection was the issue of if the
sender of the original request cannot construct a routeable SIP query,
will they be able to construct a routeable HTTP query to get to the
content indirection server. Would welcome a fuller explanation or
correction here.=20

The problem with issuing a new request falls into two parts:

-	it does not solve the problem of standalone transactions like
MESSAGE.

-	will a new request in the reverse direction reach the original
sender given that the routeing path can be totally independent of the
original request.

Again would welcome a fuller explanation or correction here.=20

Regards

Keith


> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Wednesday, July 11, 2007 4:07 PM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] Hop limit diagnostics
>=20
> For completeness - haven't we also talked about requests in=20
> the opposite direction and diagnostics by indirection? I know=20
> each of those have their own thorny problems, but the=20
> tradeoff against really large responses may make them the=20
> least unpalatable.
>=20
> And I remember conversations, but not much list traffic about=20
> a series of best-effort provisionals to carry the information.
>=20
> I still think its worth figuring out, but won't throw fits if=20
> the WG decides to put it off to the indefinite future.
>=20
> RjS
>=20
>=20
> On Jul 11, 2007, at 9:07 AM, DRAGE, Keith ((Keith)) wrote:
>=20
> > (As WG chair)
> >
> > We have a couple of related milestones on our charter that we are=20
> > stuck
> > on:
> >
> > Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS)
> > Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)
> >
> > The draft associated with this expired some way back, but=20
> you can find=20
> > it at:
> >
> > http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03
> >
> > The charter item is for a more general document that covers other=20
> > error situations as well as hop limit issues.
> >
> > However the editor's hit the intractable problem in that=20
> any transport=20
> > decision is made on the request on any particular hop, and=20
> if UDP is=20
> > used on the request, it will also be used on the response on any=20
> > particular hop. This was specified based on the assumption that any=20
> > response would not be significantly larger than the request, but as=20
> > soon as we start putting lots of useful diagnostic=20
> information in the=20
> > response, this no longer applies.
> >
> > So we are now looking for the way forward. Options include:
> >
> > A)	It is not worth the extra cycles - delete the milestone.
> >
> > B)	Limit the diagnostic information (to say around 100 bytes in the
> > worst case). If so will it contain enough useful=20
> information to make=20
> > it usable.
> >
> > C)	Solve the transport problem. And no, we do not have a debate
> > here on deprecating UDP. We've been there and done that.
> >
> > Unless people can come up with something that looks achievable, the=20
> > working group chairs are currently favouring A) above.
> >
> > Comments please.
> >
> >
> > Keith
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20


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From sip-bounces@ietf.org Thu Jul 12 07:09:37 2007
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Subject: RE: [Sip] SIP SAML IANA Consideration Sction
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>,
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Two points on this list of response code definitions:

1)	Instead of using "It is used when..." and defining the sender =
condition, can we write it if a form of the semantics the receiver =
should understand from receiving it. The sender conditions should be =
covered by normative statements along the lines of "If <condition> then =
the UAS MUST/SHOULD send a 4xx (xxx) response" elsewhere in the =
document.

2)	A question to which I do not have an answer. This defines 3 new =
response codes. Are these allocated in sufficiently generic a form. For =
example the first could be just "Assertion required of Identity header" =
and the fact that it is a SAML assertion contained in a new header =
field. I would like at some point to see some list discussion on this.

Regards

Keith

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Wednesday, July 11, 2007 10:03 AM
> To: IETF SIP List
> Subject: [Sip] SIP SAML IANA Consideration Sction
>=20
> I compiled a first version of the IANA consideration section.
> Please find it here: http://www.tschofenig.priv.at:8080/saml-sip/issue22
>=20
> Ciao
> Hannes
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] Open issue in the hitchhikers guide - seriously! Relation
	to essential corrections
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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I will be following this in the next version of =
draft-drage-sip-essential-correction unless I hear otherwise from the =
list.

Keith=20

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Wednesday, July 11, 2007 2:20 PM
> To: DRAGE, Keith (Keith)
> Cc: IETF SIP List
> Subject: Re: [Sip] Open issue in the hitchhikers guide -=20
> seriously! Relation to essential corrections
>=20
>=20
>=20
> DRAGE, Keith (Keith) wrote:
> > Your proposal as to what goes into the hitchhiker's guide sounds=20
> > sensible to me.
> >=20
> > I will take the comment about the file nameing convention into the=20
> > next version of my essential-corrections draft. Proposals=20
> for a naming=20
> > convention welcome; if no alternatives I will use something=20
> based on=20
> > what Jonathan suggests below. I see only two issues:
> >=20
> > -	we seem to be unable to decide what is a fix and what=20
> is not. If we
> > make a decision that something is a fix after the -00=20
> version, we need=20
> > to decide whether the continuity of the existing nameing=20
> has priority=20
> > or the convention of having -fix in the name.
>=20
> I think the convention has priority. SO if we decide its not=20
> a fix, it gets renamed.
>=20
> >=20
> > -	there are already a number of candidate drafts, which=20
> obviously do
> > not yet follow this convention. Again we need to decide whether to=20
> > keep the continuity of the existing naming or not.
>=20
> I favor the convention.
>=20
> -Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20


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From sip-bounces@ietf.org Thu Jul 12 09:25:11 2007
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Sean Olson" <Sean.Olson@microsoft.com>, <sip@ietf.org>
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So to explore further.=20

If the final hop is TCP and a preceding hop is UDP, does it just get
thrown away at the transport protocol boundary in this solution.

Regards

Keith

=20

> -----Original Message-----
> From: Sean Olson [mailto:Sean.Olson@microsoft.com]=20
> Sent: Wednesday, July 11, 2007 7:33 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: Hop limit diagnostics
>=20
> Why not:
>=20
> D) Define a diagnostic information mechanism that works with=20
> TCP and accept that it will not work with UDP
>=20
>=20
>=20
> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> Sent: Wednesday, July 11, 2007 7:07 AM
> To: sip@ietf.org
> Subject: [Sip] Hop limit diagnostics
>=20
> (As WG chair)
>=20
> We have a couple of related milestones on our charter that we=20
> are stuck
> on:
>=20
> Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS)
> Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)
>=20
> The draft associated with this expired some way back, but you=20
> can find it at:
>=20
> http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03
>=20
> The charter item is for a more general document that covers=20
> other error situations as well as hop limit issues.
>=20
> However the editor's hit the intractable problem in that any=20
> transport decision is made on the request on any particular=20
> hop, and if UDP is used on the request, it will also be used=20
> on the response on any particular hop. This was specified=20
> based on the assumption that any response would not be=20
> significantly larger than the request, but as soon as we=20
> start putting lots of useful diagnostic information in the=20
> response, this no longer applies.
>=20
> So we are now looking for the way forward. Options include:
>=20
> A)      It is not worth the extra cycles - delete the milestone.
>=20
> B)      Limit the diagnostic information (to say around 100=20
> bytes in the
> worst case). If so will it contain enough useful information=20
> to make it usable.
>=20
> C)      Solve the transport problem. And no, we do not have a debate
> here on deprecating UDP. We've been there and done that.
>=20
> Unless people can come up with something that looks=20
> achievable, the working group chairs are currently favouring A) above.
>=20
> Comments please.
>=20
>=20
> Keith
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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DRAGE, Keith (Keith) wrote:
> The problem with issuing a new request falls into two parts:
> 
> -	it does not solve the problem of standalone transactions like
> MESSAGE.
> 
> -	will a new request in the reverse direction reach the original
> sender given that the routeing path can be totally independent of the
> original request.
> 
> Again would welcome a fuller explanation or correction here. 

A bit of the latter was explored in S4.3 of a now expired,
though still archived draft on handling large UDP responses.
See http://tools.ietf.org/html/draft-gurbani-sip-large-udp-response-00.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
WWW:   http://www.alcatel-lucent.com/bell-labs


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--NextPart

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

	Title		: Managing Client Initiated Connections in the Session Initiation Protocol (SIP)
	Author(s)	: C. Jennings, R. Mahy
	Filename	: draft-ietf-sip-outbound-10.txt
	Pages		: 42
	Date		: 2007-7-12
	
The Session Initiation Protocol (SIP) allows proxy servers to
   initiate TCP connections and send asynchronous UDP datagrams to User
   Agents in order to deliver requests.  However, many practical
   considerations, such as the existence of firewalls and Network
   Address Translators (NATs), prevent servers from connecting to User
   Agents in this way.  This specification defines behaviors for User
   Agents, registrars and proxy servers that allow requests to be
   delivered on existing connections established by the User Agent.  It
   also defines keep alive behaviors needed to keep NAT bindings open
   and specifies the usage of multiple connections.

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   From: Volker Hilt <volkerh@bell-labs.com>

   I've submitted a new version of the "Essential Correction to the Session 
   Initiation Protocol (SIP) 503 (Service Unavailable) Response":
   http://www.ietf.org/internet-drafts/draft-hilt-sip-correction-503-01.txt

This work is important, but I don't see how it fits in the category
"essential correction", as a SIP system can function without it.
Perhaps you could add a section explaining why it is an essential
correction.

Dale


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   From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>

   Which is the appropriate 4xx message that can be used to inform user of
   an existing subscription? 

As others have said, it's not an error, so there is no error response.

   Consider the following use case: User A sends a subscribe message with a
   particular event and gets subscribed to the service. Now at a later
   point of time the user is not sure if he has subscribed for a particular
   service or not. So he might send subscribe again. So is there a way to
   inform the user that the subscription for that service for the user
   already exists?

That isn't quite the way to look at it.  Given that NOTIFYs are always
sent within a dialog, if the UA has truly forgotten about any possible
subscription, it will have also forgotten the Call-Id of the dialog,
and will have no context to recognize that an incoming NOTIFY is
relevent to its subscription.  So the old subscription must perforce
be abandoned (and it will eventually time out), and the UA must create
a new subscription.

On the other hand, if the UA knows the Call-Id of a past subscription
but doesn't know if the subscription is still valid, it can send a
re-SUBSCRIBE, and the notifier will respond 200 (if it is still valid)
or 481 (if not).

Dale


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From sip-bounces@ietf.org Thu Jul 12 14:09:29 2007
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Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>, "DRAGE,
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Subject: [Sip] Correction: WGLC for draft-ietf-sip-outbound
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Dean Willis wrote:
> I'd like to start a long working group last call for:
>
> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
>

My error. This should have referred to draft-ietf-sip-outbound-10, not
-09. It was busy working its way through the posting system and I
grabbed a pointer to the wrong one.

Sorry about that, folks.

--
Dean



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From sip-bounces@ietf.org Thu Jul 12 14:43:56 2007
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From: Sean Olson <Sean.Olson@microsoft.com>
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Date: Thu, 12 Jul 2007 11:43:11 -0700
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Subject: [Sip] RE: Hop limit diagnostics
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Yes, I think that would be appropriate.  I hate to throw out features/impro=
vement just because they don't work over UDP.

-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
Sent: Thursday, July 12, 2007 6:23 AM
To: Sean Olson; sip@ietf.org
Subject: RE: Hop limit diagnostics

So to explore further.

If the final hop is TCP and a preceding hop is UDP, does it just get
thrown away at the transport protocol boundary in this solution.

Regards

Keith



> -----Original Message-----
> From: Sean Olson [mailto:Sean.Olson@microsoft.com]
> Sent: Wednesday, July 11, 2007 7:33 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: Hop limit diagnostics
>
> Why not:
>
> D) Define a diagnostic information mechanism that works with
> TCP and accept that it will not work with UDP
>
>
>
> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> Sent: Wednesday, July 11, 2007 7:07 AM
> To: sip@ietf.org
> Subject: [Sip] Hop limit diagnostics
>
> (As WG chair)
>
> We have a couple of related milestones on our charter that we
> are stuck
> on:
>
> Jul 2007    Diagnostic Responses for SIP Errors to WGLC (PS)
> Nov 2007    Diagnostic Responses for SIP Errors to IESG (PS)
>
> The draft associated with this expired some way back, but you
> can find it at:
>
> http://tools.ietf.org/html/draft-ietf-sip-hop-limit-diagnostics-03
>
> The charter item is for a more general document that covers
> other error situations as well as hop limit issues.
>
> However the editor's hit the intractable problem in that any
> transport decision is made on the request on any particular
> hop, and if UDP is used on the request, it will also be used
> on the response on any particular hop. This was specified
> based on the assumption that any response would not be
> significantly larger than the request, but as soon as we
> start putting lots of useful diagnostic information in the
> response, this no longer applies.
>
> So we are now looking for the way forward. Options include:
>
> A)      It is not worth the extra cycles - delete the milestone.
>
> B)      Limit the diagnostic information (to say around 100
> bytes in the
> worst case). If so will it contain enough useful information
> to make it usable.
>
> C)      Solve the transport problem. And no, we do not have a debate
> here on deprecating UDP. We've been there and done that.
>
> Unless people can come up with something that looks
> achievable, the working group chairs are currently favouring A) above.
>
> Comments please.
>
>
> Keith
>
>
> _______________________________________________
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>


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Subject: RE: [Sip] SIP Identity using Media Path
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> Dan Wing Wrote:
> > > > > Section 4.1, item about Contact:
> > > > >
> > > > > How can the Contact header usefully be used in the
> > > > > signing process? An
> > > > > SBC along the message path will happily replace it.
> > > >
> > > > I have removed that in -01 (which isn't yet published, of
> > > > course).  There
> > > > was some thought it was necessary, but I agree it should be
> > > > removed from the signature.
> > > >
> > > > On a similar note, I am considering removing CallId from
> > > > the signature.
> > > > Oftentimes the Call Id value contains an IP address (in
> > > > dotted decimal
> > > > or hex), and an SBC or B2BUA may also want to rewrite such
> > > > a CallId.  I
> > > > have made a note of that in -01 so this can be discussed.
> > >
> > > I thought about that as well, but apparently I forgot about
> > > adding it to
> > > my mail. I think it would be a good idea to take Call-ID out as
> > > well.
> > 
> > Ok, thanks.
> > 
> 
> I agree with this, but keep in mind that the call-id serves as a nonce
> in RFC 4474. It might require introducing a new header/parameter to be
> the nonce.

Thanks for that reminder.  You're right, and I agree that having the
authentication service add its own nonce is the best way to do it; such
a nonce need only be a random number, as it's already qualified by the
authentication service's domain (and certificate).  Thus, SBCs won't
need to also destroy it.  I'll include that in my working revision
to sip-identity-media.

-d


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From sip-bounces@ietf.org Fri Jul 13 06:47:38 2007
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Subject: [Sip] draft-ietf-sip-consent-framework-02.txt
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Hi all,

draft-ietf-sip-consent-framework-02.txt  is not a typical framework 
document. Most framework documents I know are informational.
This document, however, seems to have the intended status "Standards Track"

Correct?

Ciao
Hannes



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From sip-bounces@ietf.org Fri Jul 13 07:02:49 2007
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(As WG chair)

I have just requested publication of:

http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-14.txt

The next stage for submitting comments is therefore the IESG last call.

The PROTO writeup submitted with the publication request is as follows.

Please feel free to respond if you consider any of the information
incorrect.

Regards

Keith


------------------------------------------------------------------------
---
PROTO writeup for
http://www.ietf.org/internet-drafts/draft-ietf-sip-consent-framework-
02.txt: "A Framework for Consent-Based Communications in the Session
Initiation Protocol=20
(SIP)"

   (1.a)  Who is the Document Shepherd for this document?  Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

Keith Drage

The document has been reviewed and is ready for forwarding to IESG for
publication.

   (1.b)  Has the document had adequate review both from key WG members
          and from key non-WG members?  Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

Document history:

-	draft-rosenberg-sipping-consent-framework-00 was submitted 8th
July 2004 and=20
expired 6th January 2005.
-	draft-ietf-sipping-consent-framework-00 was submitted 18th
October 2004 and=20
expired 18th April 2005.
-	draft-ietf-sipping-consent-framework-01 was submitted 20th
February 2005 and=20
expired 21st August 2005.
-	draft-ietf-sipping-consent-framework-02 was submitted 18th July
2005 and expired=20
19th January 2006.
-	draft-ietf-sipping-consent-framework-03 was submitted 5th
October 2005 and=20
expired 8th April 2006.
-	draft-ietf-sipping-consent-framework-04 was submitted 25th
February 2006 and=20
expired 29th August 2006.
-	draft-ietf-sipping-consent-framework-05 was submitted 12th June
2006 and expired=20
14th December 2006.
-	draft-ietf-sip-consent-framework-00 was submitted 17th September
2006 and expired=20
21st March 2007.
-	draft-ietf-sip-consent-framework-01 was submitted 26th November
2006 and expired=20
30th May 2007.
-	draft-ietf-sip-consent-framework-02 was submitted 5th July 2007
and expires 6th=20
January 2007.

WGLC was initiated in the SIP WG on draft-ietf-sip-consent-framework-00
on 25th September=20
2006 with comments requested by 17th October 2006.

Review was made and comments were received from: Jeroen van Bemmel,
Shida Schubert, Ben=20
Campbell, AC Mahendran, Mary Barnes. During the course of the work
comments have also=20
been made by: Dean Willis, Andrew Allen, Cullen Jennings, Paul Kyzivat,
Adam Roach,=20
Geoffrey Dawirs, Miguel Garcia.

The document was moved from the SIPPING WG to the SIP WG in conformance
with RFC 3427=20
because it defines new header fields and a response code. Prior review
and discussion=20
therefore took place in the SIPPING group.

Key discussions have taken place about which methods to use for various
parts of the=20
consent framework.

The document is closely related with:

-	draft-ietf-sipping-consent-format-03;
-	draft-ietf-sipping-pending-additions-02;
-	draft-ietf-sipping-uri-services-06;
-	draft-ietf-sip-uri-list-message-01;
-	draft-ietf-sip-uri-list-subscribe-01;
-	draft-ietf-sip-uri-list-conferencing-01;
-	draft-ietf-sip-multiple-refer-01.

Both OMA and 3GPP use the uri-list documents (as documented in their
PROTO writeups). As=20
these documents have a mandatory normative dependence on the consent
framework, then=20
they also need the consent framework.

   (1.c)  Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization, or XML?

The document defines mechanisms that are entirely internal to the
Session Initiation=20
Protocol (SIP). The document shepherd considers that no external review
from an external=20
specialist is necessary, apart from as follows.

While the document was generated as a result of a request from security
advisers=20
concerning the original uri-list documents (see above), the document has
not had a=20
separate security review, and that should there occur.

   (1.d)  Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of?  For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it.  In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here.  Has an IPR disclosure related to this document
          been filed?  If so, please include a reference to the
          disclosure and summarize the WG discussion and conclusion on
          this issue.

The document defines a new SIP protocol extension for a particular
purpose in a form=20
that has been used for many other extensions. The document shepherd has
no concerns with=20
the document.

There have been no IPR disclosures on this document.

   (1.e)  How solid is the WG consensus behind this document?  Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

While the document has been reviewed by appropriate SIP experts, the
level of readership=20
of the SIP working group has apparently been low. This may lead one to
assume that the=20
contents for this solution are correct, but potentially there could have
been other=20
solutions out there that have been missed.

   (1.f)  Has anyone threatened an appeal or otherwise indicated extreme
          discontent?  If so, please summarize the areas of conflict in
          separate email messages to the Responsible Area Director.  (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

None indicated.

   (1.g)  Has the Document Shepherd personally verified that the
          document satisfies all ID nits?  (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/.)  Boilerplate checks are
          not enough; this check needs to be thorough.  Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type, and URI type reviews?  If the document
          does not already indicate its intended status at the top of
          the first page, please indicate the intended status here.

The document has been reviewed against the guidelines in RFC 4485 and it
is believed=20
that the document is conformant with those guidelines.

While the document defines a new SIP response code, and two new SIP
header fields, these=20
have been performed as a SIP working group item, and therefore this
draft is in=20
conformance with RFC 3427.

For ID-NITS the checks against idnits 2.04.09 report no NITS found.

   (1.h)  Has the document split its references into normative and
          informative?  Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state?  If such normative references exist, what is the
          strategy for their completion?  Are there normative references
          that are downward references, as described in [RFC3967]?  If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].

The document has separate sections for normative and informative
references. The=20
normative references have been checked and found to be normative.

   (1.i)  Has the Document Shepherd verified that the document's IANA
          Considerations section exists and is consistent with the body
          of the document?  If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries?  Are the IANA registries clearly identified?  If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations?  Does it suggest a
          reasonable name for the new registry?  See [RFC2434].  If the
          document describes an Expert Review process, has the Document
          Shepherd conferred with the Responsible Area Director so that
          the IESG can appoint the needed Expert during IESG Evaluation?

The document defines the following values that require registration:

*	Trigger-Consent header field
*	Permission-Missing header field
*	target-uri header field parameter to Trigger-Consent header
field
*	470 response code

Section 6 of the document provides the IANA considerations section, and
this defines the=20
above.

   (1.j)  Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

The document defines two items in ABNF (Trigger-Consent and
Permission-Missing). These=20
augment the ABNF defined in RFC 3261.

Both these items pass Bill Fenner's ABNF parser in the tools webpage.

   (1.k)  The IESG approval announcement includes a Document
          Announcement Write-Up.  Please provide such a Document
          Announcement Write-Up.  Recent examples can be found in the
          "Action" announcements for approved documents.  The approval
          announcement contains the following sections:

          Technical Summary
             Relevant content can frequently be found in the abstract
             and/or introduction of the document.  If not, this may be
             an indication that there are deficiencies in the abstract
             or introduction.

          Working Group Summary
             Was there anything in the WG process that is worth noting?
             For example, was there controversy about particular points
             or were there decisions where the consensus was
             particularly rough?

          Document Quality
             Are there existing implementations of the protocol?  Have a
             significant number of vendors indicated their plan to
             implement the specification?  Are there any reviewers that
             merit special mention as having done a thorough review,
             e.g., one that resulted in important changes or a
             conclusion that the document had no substantive issues?  If
             there was a MIB Doctor, Media Type, or other Expert Review,
             what was its course (briefly)?  In the case of a Media Type
             Review, on what date was the request posted?

          Personnel
             Who is the Document Shepherd for this document?  Who is the
             Responsible Area Director?  If the document requires IANA
             experts(s), insert 'The IANA Expert(s) for the registries
             in this document are <TO BE ADDED BY THE AD>.'

Technical summary.

The Session Initiation Protocol (SIP) supports communications across
many media types,=20
including real-time audio, video, text, instant messaging, and presence.
In its current=20
form, it allows session invitations, instant messages, and other
requests to be=20
delivered from one party to another without requiring explicit consent
of the recipient. =20
Without such consent, it is possible for SIP to be used for malicious
purposes,=20
including amplification, and DoS (Denial of Service) attacks.  This
document identifies=20
a framework for consent-based communications in SIP.

Working group summary.

There is consensus in the working group to publish this document. The
document came=20
about due to security area concerns about the need to protect against
denial of service=20
attacks and amplification attacks when various relay and uri-list
mechanisms are used in=20
SIP.

Document Quality

There has been no indication of implementation.

Personnel

The document shepherd for this document was Keith Drage. The responsible
Area Director=20
was Cullen Jennings. 'The IANA Expert(s) for the registries in this
document are <TO BE=20
ADDED BY THE AD>.
----------------------------------------------------------------------


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From sip-bounces@ietf.org Fri Jul 13 07:25:39 2007
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Subject: RE: [Sip] draft-ietf-sip-consent-framework-02.txt
Date: Fri, 13 Jul 2007 13:23:35 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>,
	"IETF SIP List" <sip@ietf.org>
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The charter milestones are:

Done    Consent-Based Communications WGLC (PS) =20
Jun 2007    Consent-Based Communications to IESG (PS) =20

These quite clearly say we are working on a standards track document.

Moreover it defines certain SIP constructs for which the IANA =
registration requirements are standards track.=20

I guess as well as being the framework document that groups all the =
underlying work together, it is in its own right a SIP extension =
document.=20

As we already have 7 or 8 documents covering this issue, I suggest we =
don't perform a split to create another one. If you look at the =
dependent reference tree it is a nightmare.

Regards

Keith=20

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]=20
> Sent: Friday, July 13, 2007 11:47 AM
> To: IETF SIP List
> Subject: [Sip] draft-ietf-sip-consent-framework-02.txt
>=20
> Hi all,
>=20
> draft-ietf-sip-consent-framework-02.txt  is not a typical=20
> framework document. Most framework documents I know are informational.
> This document, however, seems to have the intended status=20
> "Standards Track"
>=20
> Correct?
>=20
> Ciao
> Hannes
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Fri Jul 13 07:35:52 2007
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I am not suggesting a document split or anything like that.

Typically, documents indicate the status.
When I open a document that says "framework" in the subject header then 
I just do not expect it todo anything else other than describing the big 
picture.

Ciao
Hannes

DRAGE, Keith (Keith) wrote:
> The charter milestones are:
>
> Done    Consent-Based Communications WGLC (PS)  
> Jun 2007    Consent-Based Communications to IESG (PS)  
>
> These quite clearly say we are working on a standards track document.
>
> Moreover it defines certain SIP constructs for which the IANA registration requirements are standards track. 
>
> I guess as well as being the framework document that groups all the underlying work together, it is in its own right a SIP extension document. 
>
> As we already have 7 or 8 documents covering this issue, I suggest we don't perform a split to create another one. If you look at the dependent reference tree it is a nightmare.
>
> Regards
>
> Keith 
>
>   
>> -----Original Message-----
>> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
>> Sent: Friday, July 13, 2007 11:47 AM
>> To: IETF SIP List
>> Subject: [Sip] draft-ietf-sip-consent-framework-02.txt
>>
>> Hi all,
>>
>> draft-ietf-sip-consent-framework-02.txt  is not a typical 
>> framework document. Most framework documents I know are informational.
>> This document, however, seems to have the intended status 
>> "Standards Track"
>>
>> Correct?
>>
>> Ciao
>> Hannes
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip 
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>     



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Thread-Topic: Publication request for draft-ietf-sip-consent-framework-02.txt
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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Subject: [Sip] RE: Publication request for
	draft-ietf-sip-consent-framework-02.txt
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And before anyone else responds on the first bit of the message below,
it should be:=20

http://www.ietf.org/internet-drafts/draft-ietf-sip-consent-framework-02.
txt

Keith=20

> -----Original Message-----
> From: DRAGE, Keith (Keith)=20
> Sent: Friday, July 13, 2007 12:01 PM
> To: sip@ietf.org
> Subject: Publication request for=20
> draft-ietf-sip-consent-framework-02.txt
>=20
> (As WG chair)
>=20
> I have just requested publication of:
>=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-gruu-14.txt
>=20
> The next stage for submitting comments is therefore the IESG=20
> last call.
>=20
> The PROTO writeup submitted with the publication request is=20
> as follows.
>=20
> Please feel free to respond if you consider any of the=20
> information incorrect.
>=20
> Regards
>=20
> Keith
>=20
>=20
> --------------------------------------------------------------
> -------------
> PROTO writeup for=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-consent-framework-
> 02.txt: "A Framework for Consent-Based Communications in the=20
> Session Initiation Protocol (SIP)"
>=20
>    (1.a)  Who is the Document Shepherd for this document?  Has the
>           Document Shepherd personally reviewed this version of the
>           document and, in particular, does he or she believe this
>           version is ready for forwarding to the IESG for publication?
>=20
> Keith Drage
>=20
> The document has been reviewed and is ready for forwarding to=20
> IESG for publication.
>=20
>    (1.b)  Has the document had adequate review both from key=20
> WG members
>           and from key non-WG members?  Does the Document=20
> Shepherd have
>           any concerns about the depth or breadth of the reviews that
>           have been performed?
>=20
> Document history:
>=20
> -	draft-rosenberg-sipping-consent-framework-00 was=20
> submitted 8th July 2004 and=20
> expired 6th January 2005.
> -	draft-ietf-sipping-consent-framework-00 was submitted=20
> 18th October 2004 and=20
> expired 18th April 2005.
> -	draft-ietf-sipping-consent-framework-01 was submitted=20
> 20th February 2005 and=20
> expired 21st August 2005.
> -	draft-ietf-sipping-consent-framework-02 was submitted=20
> 18th July 2005 and expired=20
> 19th January 2006.
> -	draft-ietf-sipping-consent-framework-03 was submitted=20
> 5th October 2005 and=20
> expired 8th April 2006.
> -	draft-ietf-sipping-consent-framework-04 was submitted=20
> 25th February 2006 and=20
> expired 29th August 2006.
> -	draft-ietf-sipping-consent-framework-05 was submitted=20
> 12th June 2006 and expired=20
> 14th December 2006.
> -	draft-ietf-sip-consent-framework-00 was submitted 17th=20
> September 2006 and expired=20
> 21st March 2007.
> -	draft-ietf-sip-consent-framework-01 was submitted 26th=20
> November 2006 and expired=20
> 30th May 2007.
> -	draft-ietf-sip-consent-framework-02 was submitted 5th=20
> July 2007 and expires 6th=20
> January 2007.
>=20
> WGLC was initiated in the SIP WG on=20
> draft-ietf-sip-consent-framework-00 on 25th September
> 2006 with comments requested by 17th October 2006.
>=20
> Review was made and comments were received from: Jeroen van=20
> Bemmel, Shida Schubert, Ben Campbell, AC Mahendran, Mary=20
> Barnes. During the course of the work comments have also been=20
> made by: Dean Willis, Andrew Allen, Cullen Jennings, Paul=20
> Kyzivat, Adam Roach, Geoffrey Dawirs, Miguel Garcia.
>=20
> The document was moved from the SIPPING WG to the SIP WG in=20
> conformance with RFC 3427 because it defines new header=20
> fields and a response code. Prior review and discussion=20
> therefore took place in the SIPPING group.
>=20
> Key discussions have taken place about which methods to use=20
> for various parts of the consent framework.
>=20
> The document is closely related with:
>=20
> -	draft-ietf-sipping-consent-format-03;
> -	draft-ietf-sipping-pending-additions-02;
> -	draft-ietf-sipping-uri-services-06;
> -	draft-ietf-sip-uri-list-message-01;
> -	draft-ietf-sip-uri-list-subscribe-01;
> -	draft-ietf-sip-uri-list-conferencing-01;
> -	draft-ietf-sip-multiple-refer-01.
>=20
> Both OMA and 3GPP use the uri-list documents (as documented=20
> in their PROTO writeups). As these documents have a mandatory=20
> normative dependence on the consent framework, then they also=20
> need the consent framework.
>=20
>    (1.c)  Does the Document Shepherd have concerns that the document
>           needs more review from a particular or broader perspective,
>           e.g., security, operational complexity, someone=20
> familiar with
>           AAA, internationalization, or XML?
>=20
> The document defines mechanisms that are entirely internal to=20
> the Session Initiation Protocol (SIP). The document shepherd=20
> considers that no external review from an external specialist=20
> is necessary, apart from as follows.
>=20
> While the document was generated as a result of a request=20
> from security advisers concerning the original uri-list=20
> documents (see above), the document has not had a separate=20
> security review, and that should there occur.
>=20
>    (1.d)  Does the Document Shepherd have any specific concerns or
>           issues with this document that the Responsible Area Director
>           and/or the IESG should be aware of?  For example, perhaps he
>           or she is uncomfortable with certain parts of the=20
> document, or
>           has concerns whether there really is a need for it.  In any
>           event, if the WG has discussed those issues and has=20
> indicated
>           that it still wishes to advance the document, detail those
>           concerns here.  Has an IPR disclosure related to=20
> this document
>           been filed?  If so, please include a reference to the
>           disclosure and summarize the WG discussion and conclusion on
>           this issue.
>=20
> The document defines a new SIP protocol extension for a=20
> particular purpose in a form that has been used for many=20
> other extensions. The document shepherd has no concerns with=20
> the document.
>=20
> There have been no IPR disclosures on this document.
>=20
>    (1.e)  How solid is the WG consensus behind this document?  Does it
>           represent the strong concurrence of a few individuals, with
>           others being silent, or does the WG as a whole=20
> understand and
>           agree with it?
>=20
> While the document has been reviewed by appropriate SIP=20
> experts, the level of readership of the SIP working group has=20
> apparently been low. This may lead one to assume that the=20
> contents for this solution are correct, but potentially there=20
> could have been other solutions out there that have been missed.
>=20
>    (1.f)  Has anyone threatened an appeal or otherwise=20
> indicated extreme
>           discontent?  If so, please summarize the areas of=20
> conflict in
>           separate email messages to the Responsible Area=20
> Director.  (It
>           should be in a separate email because this questionnaire is
>           entered into the ID Tracker.)
>=20
> None indicated.
>=20
>    (1.g)  Has the Document Shepherd personally verified that the
>           document satisfies all ID nits?  (See
>           http://www.ietf.org/ID-Checklist.html and
>           http://tools.ietf.org/tools/idnits/.)  Boilerplate=20
> checks are
>           not enough; this check needs to be thorough.  Has=20
> the document
>           met all formal review criteria it needs to, such as the MIB
>           Doctor, media type, and URI type reviews?  If the document
>           does not already indicate its intended status at the top of
>           the first page, please indicate the intended status here.
>=20
> The document has been reviewed against the guidelines in RFC=20
> 4485 and it is believed that the document is conformant with=20
> those guidelines.
>=20
> While the document defines a new SIP response code, and two=20
> new SIP header fields, these have been performed as a SIP=20
> working group item, and therefore this draft is in=20
> conformance with RFC 3427.
>=20
> For ID-NITS the checks against idnits 2.04.09 report no NITS found.
>=20
>    (1.h)  Has the document split its references into normative and
>           informative?  Are there normative references to=20
> documents that
>           are not ready for advancement or are otherwise in an unclear
>           state?  If such normative references exist, what is the
>           strategy for their completion?  Are there normative=20
> references
>           that are downward references, as described in [RFC3967]?  If
>           so, list these downward references to support the Area
>           Director in the Last Call procedure for them [RFC3967].
>=20
> The document has separate sections for normative and=20
> informative references. The normative references have been=20
> checked and found to be normative.
>=20
>    (1.i)  Has the Document Shepherd verified that the document's IANA
>           Considerations section exists and is consistent=20
> with the body
>           of the document?  If the document specifies protocol
>           extensions, are reservations requested in appropriate IANA
>           registries?  Are the IANA registries clearly identified?  If
>           the document creates a new registry, does it define the
>           proposed initial contents of the registry and an allocation
>           procedure for future registrations?  Does it suggest a
>           reasonable name for the new registry?  See=20
> [RFC2434].  If the
>           document describes an Expert Review process, has=20
> the Document
>           Shepherd conferred with the Responsible Area=20
> Director so that
>           the IESG can appoint the needed Expert during IESG=20
> Evaluation?
>=20
> The document defines the following values that require registration:
>=20
> *	Trigger-Consent header field
> *	Permission-Missing header field
> *	target-uri header field parameter to Trigger-Consent=20
> header field
> *	470 response code
>=20
> Section 6 of the document provides the IANA considerations=20
> section, and this defines the above.
>=20
>    (1.j)  Has the Document Shepherd verified that sections of the
>           document that are written in a formal language, such as XML
>           code, BNF rules, MIB definitions, etc., validate=20
> correctly in
>           an automated checker?
>=20
> The document defines two items in ABNF (Trigger-Consent and=20
> Permission-Missing). These augment the ABNF defined in RFC 3261.
>=20
> Both these items pass Bill Fenner's ABNF parser in the tools webpage.
>=20
>    (1.k)  The IESG approval announcement includes a Document
>           Announcement Write-Up.  Please provide such a Document
>           Announcement Write-Up.  Recent examples can be found in the
>           "Action" announcements for approved documents.  The approval
>           announcement contains the following sections:
>=20
>           Technical Summary
>              Relevant content can frequently be found in the abstract
>              and/or introduction of the document.  If not, this may be
>              an indication that there are deficiencies in the abstract
>              or introduction.
>=20
>           Working Group Summary
>              Was there anything in the WG process that is=20
> worth noting?
>              For example, was there controversy about=20
> particular points
>              or were there decisions where the consensus was
>              particularly rough?
>=20
>           Document Quality
>              Are there existing implementations of the=20
> protocol?  Have a
>              significant number of vendors indicated their plan to
>              implement the specification?  Are there any=20
> reviewers that
>              merit special mention as having done a thorough review,
>              e.g., one that resulted in important changes or a
>              conclusion that the document had no substantive=20
> issues?  If
>              there was a MIB Doctor, Media Type, or other=20
> Expert Review,
>              what was its course (briefly)?  In the case of a=20
> Media Type
>              Review, on what date was the request posted?
>=20
>           Personnel
>              Who is the Document Shepherd for this document? =20
> Who is the
>              Responsible Area Director?  If the document requires IANA
>              experts(s), insert 'The IANA Expert(s) for the registries
>              in this document are <TO BE ADDED BY THE AD>.'
>=20
> Technical summary.
>=20
> The Session Initiation Protocol (SIP) supports communications=20
> across many media types, including real-time audio, video,=20
> text, instant messaging, and presence.  In its current form,=20
> it allows session invitations, instant messages, and other=20
> requests to be delivered from one party to another without=20
> requiring explicit consent of the recipient. =20
> Without such consent, it is possible for SIP to be used for=20
> malicious purposes, including amplification, and DoS (Denial=20
> of Service) attacks.  This document identifies a framework=20
> for consent-based communications in SIP.
>=20
> Working group summary.
>=20
> There is consensus in the working group to publish this=20
> document. The document came about due to security area=20
> concerns about the need to protect against denial of service=20
> attacks and amplification attacks when various relay and=20
> uri-list mechanisms are used in SIP.
>=20
> Document Quality
>=20
> There has been no indication of implementation.
>=20
> Personnel
>=20
> The document shepherd for this document was Keith Drage. The=20
> responsible Area Director was Cullen Jennings. 'The IANA=20
> Expert(s) for the registries in this document are <TO BE=20
> ADDED BY THE AD>.
> --------------------------------------------------------------
> --------


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Thread-Topic: Implication of Privacy header (rfc 3323/3325) to the REFER
	message
Thread-Index: AcfFXNdunKZP0w1fQ0GNS6SCmLHTLw==
From: "Rajeev Narang (ranarang)" <ranarang@cisco.com>
To: <sip@ietf.org>
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Subject: [Sip] Implication of Privacy header (rfc 3323/3325) to the REFER
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--===============0753497909==
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This is a multi-part message in MIME format.

------_=_NextPart_001_01C7C55D.1AB798FE
Content-Type: text/plain;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

Hi,
=20
In the RFC 3323 section 4.2, the REFER message isn't listed for
implication due to Privacy header.
=20
In the following scenario, will it make sense to interpret the Privacy
header for doing the display restriction to the refer-to party.
=20
A--------
         |
B------B2BUA--------------C
=20
C sends a OOD REFER to establish a call from A to B (refer-to).
Say, REFER carries a Privacy:id header.
=20
Should B2BUA try to use the info from Privacy header in REFER header
while establishing a call from A to B. If say B is not trusted, should
B2BUA restricts the caller A's info for sending to B by not sending the
PAI header and having the From header with anonymous data.
=20
Thanks & Regards,
Rajeev
=20
=20

------_=_NextPart_001_01C7C55D.1AB798FE
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 6.00.2900.3132" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007><SPAN=20
class=3D757153114-13072007>Hi,</SPAN></SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007><SPAN=20
class=3D757153114-13072007></SPAN></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>In the =
RFC 3323=20
section 4.2, the REFER message isn't listed for implication due to =
Privacy=20
header.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>In the =
following=20
scenario, will it make sense to interpret the Privacy header for doing =
the=20
display restriction to the refer-to party.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007>A--------</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
|</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D510292314-12072007></SPAN></FONT><FONT=20
face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007>B------B2BUA--------------C</SPAN></FONT></DIV=
>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>C =
sends a OOD REFER=20
to establish a call from A to B<SPAN class=3D757153114-13072007>=20
(refer-to)</SPAN>.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>Say, =
REFER carries a=20
Privacy:id header.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D510292314-12072007><FONT face=3DArial><FONT =
size=3D2>Should B2BUA=20
try to use the info from Privacy header in REFER header while =
establishing a=20
call from A to B.<SPAN class=3D757153114-13072007> If say B is not =
trusted, should=20
B2BUA restricts the caller A's info for sending to B by not sending the =
PAI=20
header and having the From header with anonymous=20
data.</SPAN></FONT></FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>Thanks =
&amp;=20
Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007>Rajeev</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

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I think B2BUA is providing privacy service for C, so if B is not, it
should not send A's identity in Invite request to it.


________________________________

	From: Rajeev Narang (ranarang)=20
	Sent: Friday, July 13, 2007 10:48 AM
	To: sip@ietf.org
	Subject: [Sip] Implication of Privacy header (rfc 3323/3325) to
the REFERmessage
=09
=09
	Hi,
	=20
	In the RFC 3323 section 4.2, the REFER message isn't listed for
implication due to Privacy header.
	=20
	In the following scenario, will it make sense to interpret the
Privacy header for doing the display restriction to the refer-to party.
	=20
	A--------
	         |
	B------B2BUA--------------C
	=20
	C sends a OOD REFER to establish a call from A to B (refer-to).
	Say, REFER carries a Privacy:id header.
	=20
	Should B2BUA try to use the info from Privacy header in REFER
header while establishing a call from A to B. If say B is not trusted,
should B2BUA restricts the caller A's info for sending to B by not
sending the PAI header and having the From header with anonymous data.
	=20
	Thanks & Regards,
	Rajeev
	=20
	=20


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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=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3132" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D022023416-13072007>I think B2BUA is providing privacy service =
for C, so if=20
B is not, it should not send A's identity in Invite request to=20
it.</SPAN></FONT></DIV><BR>
<BLOCKQUOTE dir=3Dltr=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> Rajeev Narang (ranarang)=20
  <BR><B>Sent:</B> Friday, July 13, 2007 10:48 AM<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Implication of Privacy header =
(rfc=20
  3323/3325) to the REFERmessage<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D510292314-12072007><SPAN=20
  class=3D757153114-13072007>Hi,</SPAN></SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D510292314-12072007><SPAN=20
  class=3D757153114-13072007></SPAN></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>In =
the RFC 3323=20
  section 4.2, the REFER message isn't listed for implication due to =
Privacy=20
  header.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>In =
the following=20
  scenario, will it make sense to interpret the Privacy header for doing =
the=20
  display restriction to the refer-to party.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007>A--------</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  =
class=3D510292314-12072007>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
  |</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT><FONT face=3DArial =
size=3D2><SPAN=20
  =
class=3D510292314-12072007>B------B2BUA--------------C</SPAN></FONT></DIV=
>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>C =
sends a OOD=20
  REFER to establish a call from A to B<SPAN class=3D757153114-13072007> =

  (refer-to)</SPAN>.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D510292314-12072007>Say, =
REFER carries=20
  a Privacy:id header.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D510292314-12072007><FONT face=3DArial><FONT =
size=3D2>Should B2BUA=20
  try to use the info from Privacy header in REFER header while =
establishing a=20
  call from A to B.<SPAN class=3D757153114-13072007> If say B is not =
trusted,=20
  should B2BUA restricts the caller A's info for sending to B by not =
sending the=20
  PAI header and having the From header with anonymous=20
  data.</SPAN></FONT></FONT></SPAN></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D510292314-12072007>Thanks &amp;=20
  Regards,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007>Rajeev</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D510292314-12072007></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial =
size=3D2></FONT>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

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--===============0351537402==--





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I support this fix.

On a procedural note, its great that Robert has done the work of 
building the actual patch to be made to RFC 3261. However, I'm not sure 
if thats the form that the essential corrections spec will actually 
take. Its actually quite hard to read in this form. Indeed my main 
comment is, certainly for consideration by the working group, it would 
be better to document the actual change in a summarized form rather than 
the specific paragraph changes. That will help in our evaluation of it.

Perhaps the biggest change in here is that stray responses are no longer 
forwarded upstream; I don't think this is emphasized enough and is 
arguably a bigger problem than the request retransmit problem, which is 
more of a corner case I would say.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: "Gilad Shaham" <gshaham@juniper.net>
To: "Dan Wing" <dwing@cisco.com>,
	"Hans Persson" <hasse@ingate.com>
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Hmmm... I have seen SBCs out there that remove the authentication header
as well and not all SBCs generate authenticate challenges (for example,
they rely on a softswitch later on to do it). Also, there is the issue
of a change in Nonce in the middle of the call when a new request gets a
challenge with "stale=3Dtrue".

I'm not sure we can refrain from introducing a new value generated by
the SBC.

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Friday, July 13, 2007 3:45 AM
> To: Gilad Shaham; 'Hans Persson'
> Cc: sip@ietf.org; ietf-rtpsec@imc.org
> Subject: RE: [Sip] SIP Identity using Media Path
>=20
> > Dan Wing Wrote:
> > > > > > Section 4.1, item about Contact:
> > > > > >
> > > > > > How can the Contact header usefully be used in the
> > > > > > signing process? An
> > > > > > SBC along the message path will happily replace it.
> > > > >
> > > > > I have removed that in -01 (which isn't yet published, of
> > > > > course).  There
> > > > > was some thought it was necessary, but I agree it should be
> > > > > removed from the signature.
> > > > >
> > > > > On a similar note, I am considering removing CallId from
> > > > > the signature.
> > > > > Oftentimes the Call Id value contains an IP address (in
> > > > > dotted decimal
> > > > > or hex), and an SBC or B2BUA may also want to rewrite such
> > > > > a CallId.  I
> > > > > have made a note of that in -01 so this can be discussed.
> > > >
> > > > I thought about that as well, but apparently I forgot about
> > > > adding it to
> > > > my mail. I think it would be a good idea to take Call-ID out as
> > > > well.
> > >
> > > Ok, thanks.
> > >
> >
> > I agree with this, but keep in mind that the call-id serves as a
nonce
> > in RFC 4474. It might require introducing a new header/parameter to
be
> > the nonce.
>=20
> Thanks for that reminder.  You're right, and I agree that having the
> authentication service add its own nonce is the best way to do it;
such
> a nonce need only be a random number, as it's already qualified by the
> authentication service's domain (and certificate).  Thus, SBCs won't
> need to also destroy it.  I'll include that in my working revision
> to sip-identity-media.
>=20
> -d


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From: "Dan Wing" <dwing@cisco.com>
To: "'Gilad Shaham'" <gshaham@juniper.net>, "'Hans Persson'" <hasse@ingate.com>
Subject: RE: [Sip] SIP Identity using Media Path
Date: Fri, 13 Jul 2007 13:43:08 -0700
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> Hmmm... I have seen SBCs out there that remove the 
> authentication header

You're referring to the RFC4474 Identity and Identity-Info headers?  Or to
something else?  Yes, I fully believe and expect SBCs and many B2BUAs would
remove those.  There isn't much harm in removing them, though, especially
for an SBC, because the SBC's modification of the SDP's m/c lines will
invalidate the RFC4474 signature anyway.

> as well and not all SBCs generate authenticate challenges 
> (for example, they rely on a softswitch later on to do it).

I don't follow; what do you mean by 'generate authenticate challenge'?

> Also, there is the issue of a change in Nonce in the middle of the 
> call when a new request gets a challenge with "stale=true".

What I intend on doing in the next version is extend Identity-Fingerprints
so it contains the nonce, something like this (highlighted with "<<"):

  ...
  Identity-Fingerprints:
      nonce=1234           <<<<<<<<<<
      4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
  ...
  v=0
  ...
  m=audio 54113 RTP/SAVP 0
  a=setup:active
  a=connection:new
  a=fingerprint:SHA-1
    4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB


The contents of the Identity-Fingerprints header (which contains
the nonce generated by the authentication service and the 
a=fingerprint values from the SDP), is signed by the requestor's 
authentication service (if an enterprise, that would typically 
be your enterprise's authentication service; if you're using 
a service provider, it's your SP's authentication service).

> I'm not sure we can refrain from introducing a new value generated by
> the SBC.

-d

> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com]
> > Sent: Friday, July 13, 2007 3:45 AM
> > To: Gilad Shaham; 'Hans Persson'
> > Cc: sip@ietf.org; ietf-rtpsec@imc.org
> > Subject: RE: [Sip] SIP Identity using Media Path
> > 
> > > Dan Wing Wrote:
> > > > > > > Section 4.1, item about Contact:
> > > > > > >
> > > > > > > How can the Contact header usefully be used in the
> > > > > > > signing process? An
> > > > > > > SBC along the message path will happily replace it.
> > > > > >
> > > > > > I have removed that in -01 (which isn't yet published, of
> > > > > > course).  There
> > > > > > was some thought it was necessary, but I agree it should be
> > > > > > removed from the signature.
> > > > > >
> > > > > > On a similar note, I am considering removing CallId from
> > > > > > the signature.
> > > > > > Oftentimes the Call Id value contains an IP address (in
> > > > > > dotted decimal
> > > > > > or hex), and an SBC or B2BUA may also want to rewrite such
> > > > > > a CallId.  I
> > > > > > have made a note of that in -01 so this can be discussed.
> > > > >
> > > > > I thought about that as well, but apparently I forgot about
> > > > > adding it to
> > > > > my mail. I think it would be a good idea to take 
> Call-ID out as
> > > > > well.
> > > >
> > > > Ok, thanks.
> > > >
> > >
> > > I agree with this, but keep in mind that the call-id serves as a
> nonce
> > > in RFC 4474. It might require introducing a new 
> header/parameter to
> be
> > > the nonce.
> > 
> > Thanks for that reminder.  You're right, and I agree that having the
> > authentication service add its own nonce is the best way to do it;
> such
> > a nonce need only be a random number, as it's already 
> qualified by the
> > authentication service's domain (and certificate).  Thus, SBCs won't
> > need to also destroy it.  I'll include that in my working revision
> > to sip-identity-media.
> > 
> > -d


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jul 13 17:05:12 2007
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Subject: RE: [Sip] SIP Identity using Media Path
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	<032f01c7c58e$6b94a4c0$e3d046ab@amer.cisco.com>
From: "Gilad Shaham" <gshaham@juniper.net>
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	"Hans Persson" <hasse@ingate.com>
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Sorry, I misinterpreted what you meant in the previous response.  You
can safely ignore my last comment.

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Friday, July 13, 2007 11:43 PM
> To: Gilad Shaham; 'Hans Persson'
> Cc: sip@ietf.org; ietf-rtpsec@imc.org
> Subject: RE: [Sip] SIP Identity using Media Path
>=20
> > Hmmm... I have seen SBCs out there that remove the
> > authentication header
>=20
> You're referring to the RFC4474 Identity and Identity-Info headers?
Or to
> something else?  Yes, I fully believe and expect SBCs and many B2BUAs
> would
> remove those.  There isn't much harm in removing them, though,
especially
> for an SBC, because the SBC's modification of the SDP's m/c lines will
> invalidate the RFC4474 signature anyway.
>=20
> > as well and not all SBCs generate authenticate challenges
> > (for example, they rely on a softswitch later on to do it).
>=20
> I don't follow; what do you mean by 'generate authenticate challenge'?
>=20
> > Also, there is the issue of a change in Nonce in the middle of the
> > call when a new request gets a challenge with "stale=3Dtrue".
>=20
> What I intend on doing in the next version is extend
Identity-Fingerprints
> so it contains the nonce, something like this (highlighted with "<<"):
>=20
>   ...
>   Identity-Fingerprints:
>       nonce=3D1234           <<<<<<<<<<
>       4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
>   ...
>   v=3D0
>   ...
>   m=3Daudio 54113 RTP/SAVP 0
>   a=3Dsetup:active
>   a=3Dconnection:new
>   a=3Dfingerprint:SHA-1
>     4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
>=20
>=20
> The contents of the Identity-Fingerprints header (which contains
> the nonce generated by the authentication service and the
> a=3Dfingerprint values from the SDP), is signed by the requestor's
> authentication service (if an enterprise, that would typically
> be your enterprise's authentication service; if you're using
> a service provider, it's your SP's authentication service).
>=20
> > I'm not sure we can refrain from introducing a new value generated
by
> > the SBC.
>=20
> -d
>=20
> > > -----Original Message-----
> > > From: Dan Wing [mailto:dwing@cisco.com]
> > > Sent: Friday, July 13, 2007 3:45 AM
> > > To: Gilad Shaham; 'Hans Persson'
> > > Cc: sip@ietf.org; ietf-rtpsec@imc.org
> > > Subject: RE: [Sip] SIP Identity using Media Path
> > >
> > > > Dan Wing Wrote:
> > > > > > > > Section 4.1, item about Contact:
> > > > > > > >
> > > > > > > > How can the Contact header usefully be used in the
> > > > > > > > signing process? An
> > > > > > > > SBC along the message path will happily replace it.
> > > > > > >
> > > > > > > I have removed that in -01 (which isn't yet published, of
> > > > > > > course).  There
> > > > > > > was some thought it was necessary, but I agree it should
be
> > > > > > > removed from the signature.
> > > > > > >
> > > > > > > On a similar note, I am considering removing CallId from
> > > > > > > the signature.
> > > > > > > Oftentimes the Call Id value contains an IP address (in
> > > > > > > dotted decimal
> > > > > > > or hex), and an SBC or B2BUA may also want to rewrite such
> > > > > > > a CallId.  I
> > > > > > > have made a note of that in -01 so this can be discussed.
> > > > > >
> > > > > > I thought about that as well, but apparently I forgot about
> > > > > > adding it to
> > > > > > my mail. I think it would be a good idea to take
> > Call-ID out as
> > > > > > well.
> > > > >
> > > > > Ok, thanks.
> > > > >
> > > >
> > > > I agree with this, but keep in mind that the call-id serves as a
> > nonce
> > > > in RFC 4474. It might require introducing a new
> > header/parameter to
> > be
> > > > the nonce.
> > >
> > > Thanks for that reminder.  You're right, and I agree that having
the
> > > authentication service add its own nonce is the best way to do it;
> > such
> > > a nonce need only be a random number, as it's already
> > qualified by the
> > > authentication service's domain (and certificate).  Thus, SBCs
won't
> > > need to also destroy it.  I'll include that in my working revision
> > > to sip-identity-media.
> > >
> > > -d


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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	Here's the notes from my first pass.

NITS/editorial:

Sec 3.2
"The UA can also include a reg-id parameter which..."

Are we not assuming the UA is trying to do outbound?

Sec 4.2
"This specification assumes the set
    is determined via any of a number of configuration mechanisms, and
    future specifications can define additional mechanisms such as using
    DNS to discover this set."

Seems like this sentence is unnecessary. If you really feel the need  
to explicitly call out the fact that multiple mechanisms may exist,  
and that these mechanisms will be defined in future standards, you  
could maybe say something like:

"The mechanism or mechanisms used to configure UAs are outside the  
scope of this specification, and will be defined in future  
specifications."

If you feel the need to mention a configuration mechanism based on  
DNS, you might want to move it away from the statement that says "out  
of scope" ;)

Sec 4.2.1
It is probably obvious that third-party registrations cannot use  
outbound, but it helps the flow if you lead off with this statement.

Sec 4.3
"Unless there are
    privacy reason not to include an instance-id,"
Mixing singular and plural.

Sec 4.4
"For UDP, the upper bound of 29
       seconds was selected so that multiple STUN packets could be sent
       before 30 seconds based on information that many NATs have UDP
       timeouts as low as 30 seconds.  "

Kinda awkward. I think you could trim out the "so that multiple STUN  
packets could be sent before 30 seconds" without sacrificing clarity.

eg.
"For UDP, the upper bound of 29
       seconds was selected based on information that many NATs have UDP
       timeouts as low as 30 seconds. "

Sec 4.5
"...at least one consecutive keepalive response was received."

"...at least two consecutive keepalive responses were received."?

Sec 5.1
"   When an Edge Proxy receives a registration request with a reg-id
    header parameter in the Contact header field, it needs to determine
    if it (the edge proxy) will have to be visited for any subsequent
    requests sent to the user agent identified in the Contact header
    field, or not.  If the Edge Proxy determines that this is the case,
    it inserts its URI in a Path header field value as described in RFC
    3327 [5].  If the Edge Proxy is the first SIP node after the UAC, it
    either MUST store a "flow token"--containing information about the
    flow from the previous hop--in its Path URI, or reject the request.
    The flow token MUST be an identifier that is unique to this network
    flow.  The flow token MAY be placed in the userpart of the URI.  In
    addition, the first node MUST include an 'ob' URI parameter in its
    Path header field value.  If the Edge Proxy is not the first SIP  
node
    after the UAC it MUST NOT place an 'ob' URI parameter in a Path
    header field value.  The Edge Proxy can determine if it is the first
    hop by examining the Via header field.
"

This is a little awkward/rambling. Here's my shot at making it easier  
to read.

"   When an Edge Proxy receives a registration request with a reg-id
    header parameter in the Contact header field, it must determine
    whether it is the first SIP node after the UAC. The Edge Proxy can
    determine if it is the first hop by examining the Via header field.

    If the edge proxy is the first SIP node after the UAC, it MUST  
insert
    a Path header field value, whose URI MUST contain both a flow-token
    and an 'ob' parameter, or it MUST reject the request. The flow token
    MUST be a unique identifier for the network flow from the previous
    hop, and MAY be placed in the userpart of the URI.

    If the Edge Proxy is not the first SIP node after the UAC, it MAY
    still insert a Path header field value as described in RFC 3327 [5],
    but this value MUST NOT contain an 'ob' URI param.
    "

Sec 6, para 7
"The Registrar MUST include the 'outbound' option-tag (defined in
    Section (Section 12.1)) "

Looks like an xref is not doing what you intended.

Sec 6, para 7
"Furthermore, the Registrar MUST NOR
    include this option-tag..."

Sec 6, para 7
"Note that the requirements in this section
    applies to... "
Mixing singlar and plural.

Sec 6, para 8
"To be compliant with this specification, registrars which can receive
    SIP requests directly from a UAC without intervening edge proxies
    MUST implement STUN NAT Keepalives on its SIP UDP ports as described
    in Section 8 and when it receives a double-CRLF sequence on a
    connection oriented transport such as TCP or SCTP, it MUST
    immediately respond with a single CRLF over the same connection."

Awkward, how about:

"
Registrars that can receive SIP requests directly from a UAC (ie,
without intervening edge proxies) are required to implement the same
keepalive mechanisms as edge proxies (see Sec 5.4).
"

Sec 6, as a whole

It felt like the logic for determining whether outbound processing  
should be done was excessively spread out in this section. I could  
propose alternate text if you like.

Sec 7
"
    The proxy uses normal forwarding rules looking at the next-hop  
target
    of the message and the value of any stored Path header field vector
    in the registration binding to decide how to forward the request and
    populate the Route header in the request.  If the proxy stored
    information about the flow over which it received the REGISTER for
    the binding, then the proxy MUST send the request over the same
    'logical flow' saved with the binding that is known to deliver data
    to the specific target UA instance.
"

Awkward. How about:

"
    When the Authoritative Proxy doubles as a registrar, targets may  
have
    flow information associated with them (see Sec 6). When forwarding a
    request to a target that is associated with a flow, the proxy MUST
    send the request over said flow. Otherwise, the proxy will forward
    the request as it would normally.
"

Minor technical:

(Where should this go?)
Probably should mention that an endpoint using outbound will not be  
able to target-refresh if the edge proxy uses Record-Routes to store  
flow-tokens (provided the Dialog was established over a flow). This  
is because the edge proxy would end up routing according to flow- 
tokens in the Route-Set, which is invariant (ie, not effected by a  
target-refresh at all).

Sec 4.4.3
What if STUN requests are not answered? How many unanswered STUN  
requests must go by before the UA considers the flow dead? Will the  
occasional dropped UDP packet be sufficient evidence that the flow is  
dead?

Sec 8
"When a URI is created that refers to a SIP node that supports STUN as
    described in this section, the 'keep-stun' URI parameter, as defined
    in Section 12 MUST be added to the URI."

This can be interpreted as saying "If your server element supports  
STUN, every single URI you create that refers to it needs to have a  
keep-stun param." I don't know if this is what you intended. There  
might be valid reasons for not wanting to advertise STUN support on  
every URI you mint (indeed, not every such URI will be useful; what  
good will having 'keep-stun' in a Route header be, for example?)

Sec 8.1
"When STUN is used together with SigComp [24] compressed SIP messages
    over the same flow, how the STUN messages are sent depends on the
    transport protocol.  For UDP flows, the STUN messages are simply  
sent
    uncompressed, "outside" of SigComp."

Since STUN is only used over UDP, we can get rid of the "depends on  
transport protocol" stuff.


Major technical:

No problems.


Best regards,
Byron Campen


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; "><DIV><SPAN =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</SPAN>Here's =
the notes from my first pass.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><B>NITS/editorial:</B></DIV><=
DIV><SPAN class=3D"Apple-style-span" style=3D"text-decoration: =
underline;"><BR class=3D"khtml-block-placeholder"></SPAN></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
3.2</SPAN></DIV><DIV>"The UA can also include a reg-id parameter =
which..."</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>Are =
we not assuming the UA is trying to do outbound?</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.2</SPAN></DIV><DIV>"This specification assumes the set</DIV><DIV>=A0=A0 =
is determined via any of a number of configuration mechanisms, =
and</DIV><DIV>=A0=A0 future specifications can define additional =
mechanisms such as using</DIV><DIV>=A0=A0 DNS to discover this =
set."</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>Seems =
like this sentence is unnecessary. If you really feel the need to =
explicitly call out the fact that multiple mechanisms may exist, and =
that these mechanisms will be defined in future standards, you could =
maybe say something like:</DIV><DIV><SPAN class=3D"Apple-style-span" =
style=3D"text-decoration: underline;"><BR =
class=3D"khtml-block-placeholder"></SPAN></DIV><DIV>"The mechanism or =
mechanisms used to configure UAs are outside the scope of this =
specification, and will be defined in future =
specifications."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>If you feel the need to =
mention a configuration mechanism based on DNS, you might want to move =
it away from the statement that says "out of scope" ;)</DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;"><BR =
class=3D"khtml-block-placeholder"></SPAN></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.2.1</SPAN></DIV><DIV>It is probably obvious that third-party =
registrations cannot use outbound, but it helps the flow if you lead off =
with this statement.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.3</SPAN></DIV><DIV>"Unless there are</DIV><DIV>=A0=A0 privacy reason =
not to include an instance-id,"</DIV><DIV>Mixing singular and =
plural.</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.4</SPAN></DIV><DIV>"For UDP, the upper bound of 29</DIV><DIV>=A0 =A0 =A0=
 seconds was selected so that multiple STUN packets could be =
sent</DIV><DIV>=A0 =A0 =A0 before 30 seconds based on information that =
many NATs have UDP</DIV><DIV>=A0 =A0 =A0 timeouts as low as 30 seconds.=A0=
 "</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>Kinda =
awkward. I think you could trim out the "so that multiple STUN packets =
could be sent before 30 seconds" without sacrificing =
clarity.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>eg.</DIV><DIV>"For UDP, the =
upper bound of 29</DIV><DIV>=A0 =A0 =A0 seconds was selected based on =
information that many NATs have UDP</DIV><DIV>=A0 =A0 =A0 timeouts as =
low as 30 seconds. "</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.5</SPAN></DIV><DIV>"...at least one consecutive keepalive response was =
received."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>"...at least two =
consecutive keepalive responses were received."?</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
5.1</SPAN></DIV><DIV>"=A0=A0 When an Edge Proxy receives a registration =
request with a reg-id</DIV><DIV>=A0=A0 header parameter in the Contact =
header field, it needs to determine</DIV><DIV>=A0=A0 if it (the edge =
proxy) will have to be visited for any subsequent</DIV><DIV>=A0=A0 =
requests sent to the user agent identified in the Contact =
header</DIV><DIV>=A0=A0 field, or not.=A0 If the Edge Proxy determines =
that this is the case,</DIV><DIV>=A0=A0 it inserts its URI in a Path =
header field value as described in RFC</DIV><DIV>=A0=A0 3327 [5].=A0 If =
the Edge Proxy is the first SIP node after the UAC, it</DIV><DIV>=A0=A0 =
either MUST store a "flow token"--containing information about =
the</DIV><DIV>=A0=A0 flow from the previous hop--in its Path URI, or =
reject the request.</DIV><DIV>=A0=A0 The flow token MUST be an =
identifier that is unique to this network</DIV><DIV>=A0=A0 flow.=A0 The =
flow token MAY be placed in the userpart of the URI.=A0 In</DIV><DIV>=A0=A0=
 addition, the first node MUST include an 'ob' URI parameter in =
its</DIV><DIV>=A0=A0 Path header field value.=A0 If the Edge Proxy is =
not the first SIP node</DIV><DIV>=A0=A0 after the UAC it MUST NOT place =
an 'ob' URI parameter in a Path</DIV><DIV>=A0=A0 header field value.=A0 =
The Edge Proxy can determine if it is the first</DIV><DIV>=A0=A0 hop by =
examining the Via header field.</DIV><DIV>"</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>This is a little =
awkward/rambling. Here's my shot at making it easier to =
read.</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>"=A0=A0 =
When an Edge Proxy receives a registration request with a =
reg-id</DIV><DIV>=A0=A0 header parameter in the Contact header field, it =
must determine </DIV><DIV>=A0=A0 whether it is the first SIP node after =
the UAC. The Edge Proxy can </DIV><DIV>=A0=A0 determine if it is the =
first hop by examining the Via header field.</DIV><DIV>=A0=A0 =
</DIV><DIV>=A0=A0 If the edge proxy is the first SIP node after the UAC, =
it MUST insert</DIV><DIV>=A0=A0 a Path header field value, whose URI =
MUST contain both a flow-token </DIV><DIV>=A0=A0 and an 'ob' parameter, =
or it MUST reject the request. The flow token </DIV><DIV>=A0=A0 MUST be =
a unique identifier for the network flow from the previous </DIV><DIV>=A0=A0=
 hop, and MAY be placed in the userpart of the URI.</DIV><DIV>=A0=A0 =
</DIV><DIV>=A0=A0 If the Edge Proxy is not the first SIP node after the =
UAC, it MAY </DIV><DIV>=A0=A0 still insert a Path header field value as =
described in RFC 3327 [5], </DIV><DIV>=A0=A0 but this value MUST NOT =
contain an 'ob' URI param.</DIV><DIV>=A0=A0 "</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec 6, =
para 7</SPAN></DIV><DIV>"The Registrar MUST include the 'outbound' =
option-tag (defined in</DIV><DIV>=A0=A0 Section (Section 12.1)) =
"</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>Looks like =
an xref is not doing what you intended.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec 6, =
para 7</SPAN></DIV><DIV><SPAN class=3D"Apple-style-span">"Furthermore, =
the Registrar MUST <B>NOR</B></SPAN></DIV><DIV>=A0=A0 include this =
option-tag..."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec 6, =
para 7</SPAN></DIV><DIV>"Note that the requirements in this =
section</DIV><DIV>=A0=A0 applies to... "</DIV><DIV>Mixing singlar and =
plural.</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec 6, =
para 8</SPAN></DIV><DIV>"To be compliant with this specification, =
registrars which can receive</DIV><DIV>=A0=A0 SIP requests directly from =
a UAC without intervening edge proxies</DIV><DIV>=A0=A0 MUST implement =
STUN NAT Keepalives on its SIP UDP ports as described</DIV><DIV>=A0=A0 =
in Section 8 and when it receives a double-CRLF sequence on =
a</DIV><DIV>=A0=A0 connection oriented transport such as TCP or SCTP, it =
MUST</DIV><DIV>=A0=A0 immediately respond with a single CRLF over the =
same connection."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>Awkward, how =
about:</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>"</DIV><DIV>Registrars that =
can receive SIP requests directly from a UAC (ie, </DIV><DIV>without =
intervening edge proxies) are required to implement the same =
</DIV><DIV>keepalive mechanisms as edge proxies (see Sec =
5.4).</DIV><DIV>"</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec 6, =
as a whole</SPAN></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>It felt like the logic for =
determining whether outbound processing should be done was excessively =
spread out in this section. I could propose alternate text if you like.=A0=
 =A0 =A0=A0 =A0</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
7</SPAN></DIV><DIV>"</DIV><DIV>=A0=A0 The proxy uses normal forwarding =
rules looking at the next-hop target</DIV><DIV>=A0=A0 of the message and =
the value of any stored Path header field vector</DIV><DIV>=A0=A0 in the =
registration binding to decide how to forward the request =
and</DIV><DIV>=A0=A0 populate the Route header in the request.=A0 If the =
proxy stored</DIV><DIV>=A0=A0 information about the flow over which it =
received the REGISTER for</DIV><DIV>=A0=A0 the binding, then the proxy =
MUST send the request over the same</DIV><DIV>=A0=A0 'logical flow' =
saved with the binding that is known to deliver data</DIV><DIV>=A0=A0 to =
the specific target UA instance.</DIV><DIV>"</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>Awkward. How =
about:</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>"</DIV><DIV>=A0=A0 When the =
Authoritative Proxy doubles as a registrar, targets may have =
</DIV><DIV>=A0=A0 flow information associated with them (see Sec 6). =
When forwarding a </DIV><DIV>=A0=A0 request to a target that is =
associated with a flow, the proxy MUST </DIV><DIV>=A0=A0 send the =
request over said flow. Otherwise, the proxy will forward </DIV><DIV>=A0=A0=
 the request as it would normally.</DIV><DIV>"</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><B>Minor =
technical:</B></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>(Where should this =
go?)</DIV><DIV>Probably should mention that an endpoint using outbound =
will not be able to target-refresh if the edge proxy uses Record-Routes =
to store flow-tokens (provided the Dialog was established over a flow). =
This is because the edge proxy would end up routing according to =
flow-tokens in the Route-Set, which is invariant (ie, not effected by a =
target-refresh at all).</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
4.4.3</SPAN></DIV><DIV>What if STUN requests are not answered? How many =
unanswered STUN requests must go by before the UA considers the flow =
dead? Will the occasional dropped UDP packet be sufficient evidence that =
the flow is dead?</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
8</SPAN></DIV><DIV>"When a URI is created that refers to a SIP node that =
supports STUN as</DIV><DIV>=A0=A0 described in this section, the =
'keep-stun' URI parameter, as defined</DIV><DIV>=A0=A0 in Section 12 =
MUST be added to the URI."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>This can be interpreted as =
saying "If your server element supports STUN, every single URI you =
create that refers to it needs to have a keep-stun param." I don't know =
if this is what you intended. There might be valid reasons for not =
wanting to advertise STUN support on every URI you mint (indeed, not =
every such URI will be useful; what good will having 'keep-stun' in a =
Route header be, for example?)</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><SPAN =
class=3D"Apple-style-span" style=3D"text-decoration: underline;">Sec =
8.1</SPAN></DIV><DIV>"When STUN is used together with SigComp [24] =
compressed SIP messages</DIV><DIV>=A0=A0 over the same flow, how the =
STUN messages are sent depends on the</DIV><DIV>=A0=A0 transport =
protocol.=A0 For UDP flows, the STUN messages are simply =
sent</DIV><DIV>=A0=A0 uncompressed, "outside" of SigComp."</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>Since STUN is only used =
over UDP, we can get rid of the "depends on transport protocol" =
stuff.</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><B>Major =
technical:</B></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>No problems.</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>Best =
regards,</DIV><DIV>Byron Campen</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV></BODY></HTML>=

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Subject: [Sip] SIP Location Conveyance-08 is available 
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SIP WG

I have updated Location Conveyance to -08.

http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-08.txt

Venture below for the list of changes, already know issues with -08, 
and what I know I'm adding (unless told otherwise  ;-)

>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           : Location Conveyance for the Session 
> Initiation Protocol
>         Author(s)       : J. Polk, B. Rosen
>         Filename        : draft-ietf-sip-location-conveyance-08.txt
>         Pages           : 38
>         Date            : 2007-7-11
>
>This document defines an extension to the Session Initiation
>    Protocol (SIP) to convey geographic location information from one
>    SIP entity to another SIP entity.  The extension covers end to end
>    conveyance as well as location-based routing, where proxy servers
>    make routing decisions based on the location of the SIP user agents.

The following is a brief listing of the (numerous) changes made:

- made clear the original intention of the doc to show that this is 
for sending location of a Target, and not limited to sending location 
of the UAC, as I've read in several messages to the various lists 
about this ID's purpose.  Each PIDF-LO has an identifier in it.  This 
means more than one PIDF-LO can be in the same SIP request *but* each 
can be for their own Target. This means more than one Target location 
can be in the same request.  That said, only one Target can be in one 
PIDF-LO. Meaning, all location information contained within a single 
PIDF-LO must point at the location of the identified Target of that 
PIDF-LO, regardless of the internal formats used within the 
PIDF-LO.  This allows, for example, the mixing of civic and 
geo/coordinate location XML elements - because they all apply to the 
ONE location of the Target.

- I cleaned up the text to make Target, Location Recipient, Location 
Server, Location Object consistent.

- I changed references from "routing" a message to "retargeting" a 
message whenever the location in a request was used to determine 
where to forward the message (by a SIP server).

- Changed error reporting in a response from a Warning code to a new 
Geolocation-Error header in a  (meaning any) response.  Kept the 424 
(Bad Location Information) error for times in which a request's 
included location was the reason the request was rejected.

- Added text stating that a transaction can be either successful or 
have another response (other than 424) and still have a 
Geolocation-Error header.

- added a Geolocation header parameter "recipient=" to indication 
which SIP entity type a particular locationValue is intended for.

         OPEN ISSUE here: what does a Location Recipient
                 do if more than one locationValue point to it? (i.e.,
                 the case in which the UAS receives 2 locations
                 and both state "recipient=endpoint")

- elaborated on the existing header parameters within the Geolocation 
header "inserted-by=", "used-for-routing" (and the new "recipient=" 
parameter). The text makes it clearer than any or all 3 of these of 
these parameters can be present in a locationValue.

- expanded the list of Methods that can optionally have the 
Geolocation header to every one *but* ACK and CANCEL. I left off 
PRACK from the list, but that was by mistake - and will be corrected in -09.

- Geolocation-Error can appear in the same Methods as the Geolocation header.

- there can be one or more locationErrorValues in a response, 
allowing each location in a request to be identified as have bad 
parts, and giving the Location Recipient the ability to telling each 
location inserter what was wrong with their inserted location.

- because each locationErrorValue in a response has exactly one 
"inserter=" entity identified, this creates the ability for each 
response receiver to know of a particular locationErrorValue was 
meant for them, and to ignore all others.  This solves the spoken 
requirement of if an error is received, how does the receiver know 
that response was meant for them (since more than one entity can 
insert location into the request along the path).

- The ID RECOMMENDS now that a second locationValue not be added to a 
request that already has one for a given Target. This will lead to 
confusion by the Location Recipient.  There has been no consensus 
what to do if this situation would occur - though many have voiced 
opinions this should be addressed.  The only way I know how to 
address this is either to prioritize the locationValues based on 
which entity inserted them (trusting one over the other).  For 
example, trusting a UA inserter over a Proxy inserter.  This cuts 
into the concept of Geopriv Sighting (RFC3693).  What about trusting 
LbyR over LbyV. Unfortunately this doesn't give any notion of which 
entity inserted either location type...

- clarified that the "used-for-routing" header parameter of a 
locationValue is to be kept in the locationValue even if a subsequent 
proxy routes the request again, but on a different locationValue 
within the request.  This will help a Location Recipient determine 
how the request got to this destination (used more for after the fact 
troubleshooting).

         OPEN ISSUE here: I added text here stating something unusual in SIP,
                      the concept of inserting header values in a 
certain order to
                      ensure a path of hops could be 
determined.  Robert Sparks has
                      already pointed out there is another, more consistent way
                      to solve this. I will take his direction to be 
more consistent here...

- Clarified the rough consensus stance that location not be carried 
in any response.  This is for fragmentations reasons, as well as the 
inability to error that locationValue.

- increased the number of IANA registrations to include the 
locationValue header parameters, and given brief definitions to the 
Geolocation-Error codes (similar to what Warning has); also 
registered the new Geolocation-Error header.

- Appendix B was all the "changes since last version". It is now an 
INVITE request with an S/MIME protected PIDF-LO using the civic 
format.  This location position is the same as the (x/y) Coordinate 
position in section 4.1.

- plus cleaned up a bunch of other little nits throughout doc



What I already know is missing or in error in -08 (that will be fixed in -09)

- there is no event package for the fetch function the SUB/NOT to 
dereference an initial PIDF-LO of the Target.  This event package 
will be (something like) location-fetch, for one-time only fetches of 
location of the Target, not for pending subscriptions or location 
tracking or anything like that.  Someone wanting that capability will 
need to wrote a new doc for that, cuz it's not going in this doc.

- I missed changing the Geolocation-Error examples from the format of 
the Warning header, that used SPACE between header values, where the 
BNF of the Geolocation-Error called for a SEMI between values. This 
is an artifact from changing this error reporting from Reason to 
Warning to a new header.... I got the 1st example right, then missed 
the remaining ones (oops!)

- Need to fix the BNF of the Geolocation-Error header wrt to the host 
name - to be equivalent to the "sent-by" FQDN of the Via, identifying 
the location inserting entity of a particular locationValue (that had 
an error in it).

- I need to get a solution idea for and then consensus of the two 
OPEN ISSUES listed above...


What I'm planning on adding (until rough consensus tells me not to) 
is the following:

- the ability for the Geolocation-Error locationErrorValue to contain 
each CAType that the Location Recipient had a problem with (from the 
request).  The CAType values will *NOT* be sent, just the CATypes 
themselves.  This greatly adds to the information about the location 
error experienced by the Location Recipient (to the Location 
inserter) in the response.

- I might break out the text regarding the number of locations in a 
request, and what those limitations are; because I've had a lot of 
discussion about how this facet is interpreted - and more than a few 
don't get it right - meaning I'm not being clear enough.  This means 
there is a lack of clarity in the text.  A separate subsection might 
be necessary.

comments (or questions) are appreciated

James




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SIP WG

I've updated this ID for your consideration.

I've cleaned up the number of new namespaces, as well as the number 
of priority-values per namespace.

This is a very simple IANA registration document with a very small 
update to RFC 4412 included, which is adding a delimiter in RPH 
namespaces to enable sub-namespaces.

The requirement for this is old from a very large customer, and has 
already been codified by most (all?) willing vendors of this customer.

Comments are appreciated

James

>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespaces-01.txt
>
>         Title           : New Session Initiation Protocol 
> Resource-Priority Header Namespaces for the Defense Information Systems Agency
>         Author(s)       : J. Polk
>         Filename        : draft-polk-sip-rph-new-namespaces-01.txt
>         Pages           : 12
>         Date            : 2007-7-13
>
>This document creates additional Session Initiation Protocol
>    Resource-Priority header namespaces, to be IANA registered.  This
>    document intends to update RFC 4412, as a Proposed Standard document
>    if published by the RFC-Editor.


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Mainly editorial comments.

It is a little  confusing that section 2.1 refers to
"dsn-usarmy
 dsn-usnavy
 dsn-usairforce
 dsn-usmarines"

but those are not among the new namespaces being created in this document. 
 It would be clearer if you used the actual proposed namespaces (e.g.,
dsn-000000      dsn-000010      dsn-000020      dsn-000030) in the 
discussion of section 2.1.

When I was explaining the earlier version to someone, we got off on an 
extended tangent about whether or not DISA "would separate army calls from 
navy calls".  If the example  used the numerical "sub domains", that 
confusion would be avoided.

If I am understanding correctly, there is (or a least may be) a one-to-one 
correspondence between the "domain-subdomain" as defined here, and the 
"domain-subdomain"  defined in draft-polk-tsvwg-signaled-domain-id-00. 
draft-polk-tsvwg-signaled-domain-id-00 mentions this relationship.  Would 
it make sense to mention the relationship here as well?

Janet


"James M. Polk" <jmpolk@cisco.com> wrote on 07/13/2007 10:26:54 PM:

> SIP WG
> 
> I've updated this ID for your consideration.
> 
> I've cleaned up the number of new namespaces, as well as the number 
> of priority-values per namespace.
> 
> This is a very simple IANA registration document with a very small 
> update to RFC 4412 included, which is adding a delimiter in RPH 
> namespaces to enable sub-namespaces.
> 
> The requirement for this is old from a very large customer, and has 
> already been codified by most (all?) willing vendors of this customer.
> 
> Comments are appreciated
> 
> James
> 
> >A URL for this Internet-Draft is:
> 
>http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespaces-01.txt
> >
> >         Title           : New Session Initiation Protocol 
> > Resource-Priority Header Namespaces for the Defense Information 
> Systems Agency
> >         Author(s)       : J. Polk
> >         Filename        : draft-polk-sip-rph-new-namespaces-01.txt
> >         Pages           : 12
> >         Date            : 2007-7-13
> >
> >This document creates additional Session Initiation Protocol
> >    Resource-Priority header namespaces, to be IANA registered.  This
> >    document intends to update RFC 4412, as a Proposed Standard 
document
> >    if published by the RFC-Editor.
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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<br><font size=2 face="sans-serif">Mainly editorial comments.</font>
<br>
<br><font size=2 face="sans-serif">It is a little &nbsp;confusing that
section 2.1 refers to</font>
<br><font size=2 face="sans-serif">&quot;</font><font size=3><tt>dsn-usarmy<br>
 dsn-usnavy<br>
 dsn-usairforce<br>
 dsn-usmarines&quot;<br>
<br>
</tt></font><font size=2 face="sans-serif">but those are not among the
new namespaces being created in this document. &nbsp;It would be clearer
if you used the actual proposed namespaces (e.g.,</font><font size=3><tt>dsn-000000
&nbsp; &nbsp; &nbsp;dsn-000010 &nbsp; &nbsp; &nbsp;dsn-000020 &nbsp; &nbsp;
&nbsp;dsn-000030</tt></font><font size=2 face="sans-serif">) in the discussion
of section 2.1.</font>
<br>
<br><font size=2 face="sans-serif">When I was explaining the earlier version
to someone, we got off on an extended tangent about whether or not DISA
&quot;would separate army calls from navy calls&quot;. &nbsp;If the example
&nbsp;used the numerical &quot;sub domains&quot;, that confusion would
be avoided.</font>
<br>
<br><font size=2 face="sans-serif">If I am understanding correctly, there
is (or a least may be) a one-to-one correspondence between the &quot;domain-subdomain&quot;
as defined here, and the &quot;domain-subdomain&quot; &nbsp;defined in
draft-polk-tsvwg-signaled-domain-id-00. &nbsp;draft-polk-tsvwg-signaled-domain-id-00
mentions this relationship. &nbsp;Would it make sense to mention the relationship
here as well?</font>
<br>
<br><font size=2 face="sans-serif">Janet<br>
</font>
<br>
<br><font size=2><tt>&quot;James M. Polk&quot; &lt;jmpolk@cisco.com&gt;
wrote on 07/13/2007 10:26:54 PM:<br>
<br>
&gt; SIP WG<br>
&gt; <br>
&gt; I've updated this ID for your consideration.<br>
&gt; <br>
&gt; I've cleaned up the number of new namespaces, as well as the number
<br>
&gt; of priority-values per namespace.<br>
&gt; <br>
&gt; This is a very simple IANA registration document with a very small
<br>
&gt; update to RFC 4412 included, which is adding a delimiter in RPH <br>
&gt; namespaces to enable sub-namespaces.<br>
&gt; <br>
&gt; The requirement for this is old from a very large customer, and has
<br>
&gt; already been codified by most (all?) willing vendors of this customer.<br>
&gt; <br>
&gt; Comments are appreciated<br>
&gt; <br>
&gt; James<br>
&gt; <br>
&gt; &gt;A URL for this Internet-Draft is:<br>
&gt; &gt;http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespaces-01.txt<br>
&gt; &gt;<br>
&gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; Title &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; : New Session Initiation Protocol <br>
&gt; &gt; Resource-Priority Header Namespaces for the Defense Information
<br>
&gt; Systems Agency<br>
&gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; Author(s) &nbsp; &nbsp; &nbsp; :
J. Polk<br>
&gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;:
draft-polk-sip-rph-new-namespaces-01.txt<br>
&gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; : 12<br>
&gt; &gt; &nbsp; &nbsp; &nbsp; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;: 2007-7-13<br>
&gt; &gt;<br>
&gt; &gt;This document creates additional Session Initiation Protocol<br>
&gt; &gt; &nbsp; &nbsp;Resource-Priority header namespaces, to be IANA
registered. &nbsp;This<br>
&gt; &gt; &nbsp; &nbsp;document intends to update RFC 4412, as a Proposed
Standard document<br>
&gt; &gt; &nbsp; &nbsp;if published by the RFC-Editor.<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
</tt></font>
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Subject: Re: [Sip] Update to draft-polk-sip-rph-new-namespaces-01.txt
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At 11:35 PM 7/13/2007, Janet P Gunn wrote:

>Mainly editorial comments.
>
>It is a little  confusing that section 2.1 refers to
>"dsn-usarmy
>dsn-usnavy
>dsn-usairforce
>dsn-usmarines"
>
>but those are not among the new namespaces being created in this 
>document.  It would be clearer if you used the actual proposed 
>namespaces 
>(e.g.,dsn-000000      dsn-000010      dsn-000020      dsn-000030) in 
>the discussion of section 2.1.

I suppose I could use example that are being registered, but the idea 
of a division of a domain by recognized service boundaries is 
something that I thought would help readers understand the concept better with.


>When I was explaining the earlier version to someone, we got off on 
>an extended tangent about whether or not DISA "would separate army 
>calls from navy calls".

well.... they have had us vendors plan and codify for this for more 
than 2 years, so it's a reality.

>If the example  used the numerical "sub domains", that confusion 
>would be avoided.

what would you change the description to?

DSN is a domain. Divide it up into several subsets of domains and one 
c/would logically come up with the term sub-domain -- mean part of 
the larger, which each is in this case.


>If I am understanding correctly, there is (or a least may be) a 
>one-to-one correspondence between the "domain-subdomain" as defined 
>here, and the "domain-subdomain"  defined in 
>draft-polk-tsvwg-signaled-domain-id-00.

this *can* be a configuration, but it does not have to be - so this 
relationship (as you mention) isn't a hard-and-fast rule to be 
universally applied to either doc.

>draft-polk-tsvwg-signaled-domain-id-00 mentions this 
>relationship.  Would it make sense to mention the relationship here as well?

I can mention draft-polk-tsvwg-signaled-domain-id-00 as an 
informative reference in this doc, yes.

James


>Janet
>
>
>"James M. Polk" <jmpolk@cisco.com> wrote on 07/13/2007 10:26:54 PM:
>
> > SIP WG
> >
> > I've updated this ID for your consideration.
> >
> > I've cleaned up the number of new namespaces, as well as the number
> > of priority-values per namespace.
> >
> > This is a very simple IANA registration document with a very small
> > update to RFC 4412 included, which is adding a delimiter in RPH
> > namespaces to enable sub-namespaces.
> >
> > The requirement for this is old from a very large customer, and has
> > already been codified by most (all?) willing vendors of this customer.
> >
> > Comments are appreciated
> >
> > James
> >
> > >A URL for this Internet-Draft is:
> > >http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namesp 
> aces-01.txt
> > >
> > >         Title           : New Session Initiation Protocol
> > > Resource-Priority Header Namespaces for the Defense Information
> > Systems Agency
> > >         Author(s)       : J. Polk
> > >         Filename        : draft-polk-sip-rph-new-namespaces-01.txt
> > >         Pages           : 12
> > >         Date            : 2007-7-13
> > >
> > >This document creates additional Session Initiation Protocol
> > >    Resource-Priority header namespaces, to be IANA registered.  This
> > >    document intends to update RFC 4412, as a Proposed Standard document
> > >    if published by the RFC-Editor.
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip


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"James M. Polk" <jmpolk@cisco.com> wrote on 07/15/2007 06:21:06 PM:

> At 11:35 PM 7/13/2007, Janet P Gunn wrote:
> 
> >Mainly editorial comments.
> >
> >It is a little  confusing that section 2.1 refers to
> >"dsn-usarmy
> >dsn-usnavy
> >dsn-usairforce
> >dsn-usmarines"
> >
> >but those are not among the new namespaces being created in this 
> >document.  It would be clearer if you used the actual proposed 
> >namespaces 
> >(e.g.,dsn-000000      dsn-000010      dsn-000020      dsn-000030) in 
> >the discussion of section 2.1.
> 
...
> 
> what would you change the description to?
> 
> DSN is a domain. Divide it up into several subsets of domains and one 
> c/would logically come up with the term sub-domain -- mean part of 
> the larger, which each is in this case.
 
Probably, I would just add a sentence  saying something like 
"These examples illustrate the principle of the domain and subdomain 
within a namespace but are not the actual namespaces being registered."

Janet
--=_alternative 007E2F8485257319_=
Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif"><br>
</font>
<br>
<br><font size=2><tt>&quot;James M. Polk&quot; &lt;jmpolk@cisco.com&gt;
wrote on 07/15/2007 06:21:06 PM:<br>
<br>
&gt; At 11:35 PM 7/13/2007, Janet P Gunn wrote:<br>
&gt; <br>
&gt; &gt;Mainly editorial comments.<br>
&gt; &gt;<br>
&gt; &gt;It is a little &nbsp;confusing that section 2.1 refers to<br>
&gt; &gt;&quot;dsn-usarmy<br>
&gt; &gt;dsn-usnavy<br>
&gt; &gt;dsn-usairforce<br>
&gt; &gt;dsn-usmarines&quot;<br>
&gt; &gt;<br>
&gt; &gt;but those are not among the new namespaces being created in this
<br>
&gt; &gt;document. &nbsp;It would be clearer if you used the actual proposed
<br>
&gt; &gt;namespaces <br>
&gt; &gt;(e.g.,dsn-000000 &nbsp; &nbsp; &nbsp;dsn-000010 &nbsp; &nbsp;
&nbsp;dsn-000020 &nbsp; &nbsp; &nbsp;dsn-000030) in <br>
&gt; &gt;the discussion of section 2.1.<br>
&gt; <br>
...<br>
&gt; <br>
&gt; what would you change the description to?<br>
&gt; <br>
&gt; DSN is a domain. Divide it up into several subsets of domains and
one <br>
&gt; c/would logically come up with the term sub-domain -- mean part of
<br>
&gt; the larger, which each is in this case.<br>
 </tt></font>
<br><font size=2><tt>Probably, I would just add a sentence &nbsp;saying
something like </tt></font>
<br><font size=2><tt>&quot;These examples illustrate the principle of the
domain and subdomain within a namespace but are not the actual namespaces
being registered.&quot;</tt></font>
<br>
<br><font size=2><tt>Janet</tt></font>
--=_alternative 007E2F8485257319_=--



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Date: Sun, 15 Jul 2007 18:17:55 -0500
To: Janet P Gunn <jgunn6@csc.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] Update to draft-polk-sip-rph-new-namespaces-01.txt
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At 05:58 PM 7/15/2007, Janet P Gunn wrote:
>"James M. Polk" <jmpolk@cisco.com> wrote on 07/15/2007 06:21:06 PM:
>
> > At 11:35 PM 7/13/2007, Janet P Gunn wrote:
> >
> > >Mainly editorial comments.
> > >
> > >It is a little  confusing that section 2.1 refers to
> > >"dsn-usarmy
> > >dsn-usnavy
> > >dsn-usairforce
> > >dsn-usmarines"
> > >
> > >but those are not among the new namespaces being created in this
> > >document.  It would be clearer if you used the actual proposed
> > >namespaces
> > >(e.g.,dsn-000000      dsn-000010      dsn-000020      dsn-000030) in
> > >the discussion of section 2.1.
> >
>...
> >
> > what would you change the description to?
> >
> > DSN is a domain. Divide it up into several subsets of domains and one
> > c/would logically come up with the term sub-domain -- mean part of
> > the larger, which each is in this case.
>
>Probably, I would just add a sentence  saying something like
>"These examples illustrate the principle of the domain and subdomain 
>within a namespace but are not the actual namespaces being registered."

something like this is easy to add, so I will.


>Janet


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From sip-bounces@ietf.org Mon Jul 16 04:20:41 2007
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Date: Mon, 16 Jul 2007 16:19:15 +0800
From: Qian Sun <sunqian@huawei.com>
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Hi,

As this draft mentioned, the presence rules may affect the views of the resource state. If the presence rules of presentity are modified, all of etags for this presentity have to be abandoned. Or the PS has to evaluate whether each etag still is valid. 

For filtering, there is similar problem:
http://www1.ietf.org/mail-archive/web/sip/current/msg19724.html

These are difference between HTTP and SIP events framework.

Cheers,
Qian





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From sip-bounces@ietf.org Mon Jul 16 04:29:13 2007
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Subject: RE: [Sip] UA Loose Routing
Date: Mon, 16 Jul 2007 10:25:53 +0200
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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Hi,=20

>>>My objective was to reduce this count. Indeed I was hopeful that=20
>>>UA-loose-route would allow us to deprecate P-Called-Party-ID, not the

>>>other way around. WIth UA-loose-route, we end up with:
>>>=20
>>>Request-URI: identifies the target Route: tells you how to get to the

>>>target To: tells you the original target History info: sequence of=20
>>>identities of targets
>>=20
>>If people can't explain to their implementors the meaning of different

>>SIP headers maybe THAT is something we should concentrate on, and if=20
>>needed write a draft about.
>=20
>Its not a question of documentation. Its that we would in=20
>fact have multiple headers which mean the same thing.

So, for my clarification, which headers would mean the same thing using
P-Called-Party-ID?

>>>which is much cleaner.
>>=20
>>If we would re-do SIP I am sure lots of things could be done cleaner.
>>=20
>>=20
>>P-Called-Party-ID does not change the current routing mechanism, and=20
>>it does not have the potential issue with proxies (and possible also
>>SBCs) that you have mentioned. And the proxy re-writing the R-URI can=20
>>use the mechanism no matter if the UA supports it or not.
>=20
>Sure, but its useless unless the UA supports it. So my point=20
>is, in all cases, we are talking about changes to UA and to=20
>proxy to do something useful here. The UA-loose-route=20
>mechanism doesn't require any changes beyond those two.

The P-Called-Party-ID solution doesn't require any changes to proxies at
all.

And, even if a proxy wouldn't have a problem with the loose routing
solution as such, in some environments the proxies processes the
P-Called-Party-ID header. So, you would still need to update those
proxies, if you want to replace the P- header mechanism with something
else.
=20
>>>So, functionally, P-Called-Party-ID could work. Any number of things
*can* work.
>>=20
>>Yes, but P-Called-Party-ID is a standardized mechanism, made for this
purpose, which most likely does work.
>=20
>We have lots of P-headers that already exist. That doesn't mean we
shouldn't bother to provide an architecturally better=20
>solution.

I don't think a solution based on a P- header automatically is a "bad"
solution. In this case the main purpose is to provide a URI, and a
header (whether it's a P- header or not is not relevant) is perfectly ok
for that I think.

And, in general I think we give the market the wrong picture if we make
changes that may affect existing deployments and systems (the
P-Called-Party header has been part of IMS for a realtively long time) -
especailly when nothing is broken with the current systems.

Yes, maybe there are things we think could have been done cleaner, but
in some cases we just have to accept the fact that things have been
defined in a specific way.

Regards,

Christer





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Subject: RE: [Sip] SIP Identity using Media Path
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Dan Wing" <dwing@cisco.com>, "Hans Persson" <hasse@ingate.com>,
	<sip@ietf.org>
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Hi Dan,=20

>>Section 4.1, item about Contact:
>>=20
>>How can the Contact header usefully be used in the signing process? An

>>SBC along the message path will happily replace it.
>=20
>I have removed that in -01 (which isn't yet published, of=20
>course).  There was some thought it was necessary, but I=20
>agree it should be removed from the signature.
>=20
>On a similar note, I am considering removing CallId from the=20
>signature.
>Oftentimes the Call Id value contains an IP address (in=20
>dotted decimal or hex), and an SBC or B2BUA may also want to=20
>rewrite such a CallId.  I have made a note of that in -01 so=20
>this can be discussed.

An SBC can modify almost any header. Looking at chapter 4.1, I would say
that at least the To- and From headers are in the "risk zone" (EVEN if
you only use the addr-spec part).

The CSeq (at least the digit portion) may also be modified, for example
if there has been some "dialog piggybacked" requests sent between the
SBC and another entity, but not end-to-end. In that case the the SBC may
have to increase the CSeq before forwarding the request, if the digit
portion value has already been used in a request sent by the SBC in the
same direction.

Regards,

Christer


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From sip-bounces@ietf.org Mon Jul 16 10:15:04 2007
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Hi Guys,

I am facing a problem in following scenario on WinCE 5.0. Please let me know if you got an answer for this ASAP.

In SIP Client
1.  I have opened server TCP and UDP on 5060 port for polling responses aswell as request
2.  To send a SIP packet, I have opened a TCP client socket 5060 for destination. I got a socket descriptor as 9

SIP request goes on dest port as 5060, src port as 1997.
Server is responding on src port 1997. Hence my polling is failing as it is for TCP and UDP sockets.

Src port 1997 is not constant. It is just random. for me to poll, I need socket descriptor. How to get this logical port number given TCP? If I can get this, I can create a socket and get the descriptor and add to polling list

Waiting a quick response if possible.

Thanks !
Rathod.




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From sip-bounces@ietf.org Mon Jul 16 10:41:53 2007
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If you got socket descriptor 9 when creating the socket, and to
poll you need the socket descriptor (and if you really need to poll)
then poll socket descriptor 9.
Or is this not possible?


Regards,
Attila

=20

-----Original Message-----
From: Rathod Subhashchandra [mailto:rathod@tataelxsi.co.in]=20
Sent: 16 July 2007 15:15
To: sip@ietf.org
Cc: ietf-rtpsec@imc.org
Subject: [Sip] Sending SIP packet over TCP

Hi Guys,

I am facing a problem in following scenario on WinCE 5.0. Please let me
know if you got an answer for this ASAP.

In SIP Client
1.  I have opened server TCP and UDP on 5060 port for polling responses
aswell as request 2.  To send a SIP packet, I have opened a TCP client
socket 5060 for destination. I got a socket descriptor as 9

SIP request goes on dest port as 5060, src port as 1997.
Server is responding on src port 1997. Hence my polling is failing as it
is for TCP and UDP sockets.

Src port 1997 is not constant. It is just random. for me to poll, I need
socket descriptor. How to get this logical port number given TCP? If I
can get this, I can create a socket and get the descriptor and add to
polling list

Waiting a quick response if possible.

Thanks !
Rathod.




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


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Subject: RE: [Sip] SIP Identity using Media Path
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> An SBC can modify almost any header.

Yes.

> Looking at chapter 4.1, 
> I would say that at least the To- and From headers are in 
> the "risk zone" (EVEN if you only use the addr-spec part).

If the From header is modified, the identity is broken 
anyway.  Such modifications of the From header are discussed
in the Background section of draft-wing-sip-identity-media-00.

Allowing modifications of the To header creates the 
opportunity for an attacker to launch a replay attack by
substituting the To address and replaying the message 
until the Date: header is old enough that receiving 
authentication servers reject the Date: header out of 
hand.

> The CSeq (at least the digit portion) may also be modified, 
> for example if there has been some "dialog piggybacked" requests 
> sent between the SBC and another entity, but not end-to-end. In 
> that case the the SBC may have to increase the CSeq before 
> forwarding the request, if the digit portion value has already 
> been used in a request sent by the SBC in the same direction.

That seems like a good opportunity, within that trust domain 
between the SBC and the other entity, to use 
P-Asserted-Identity.

-d


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From sip-bounces@ietf.org Mon Jul 16 13:24:04 2007
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Subject: RE: [Sip] SIP Identity using Media Path
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Hi,

>>The CSeq (at least the digit portion) may also be modified,
>>for example if there has been some "dialog piggybacked" requests
>>sent between the SBC and another entity, but not end-to-end. In
>>that case the the SBC may have to increase the CSeq before
>>forwarding the request, if the digit portion value has already
>>been used in a request sent by the SBC in the same direction.
>
>That seems like a good opportunity, within that trust domain
>between the SBC and the other entity, to use
>P-Asserted-Identity.

I am not sure I understand. How could the P-A-I header be used instead =
of the CSeq?

Regards,
=20
Christer
=20

=20



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> >>The CSeq (at least the digit portion) may also be modified,
> >>for example if there has been some "dialog piggybacked" requests
> >>sent between the SBC and another entity, but not end-to-end. In
> >>that case the the SBC may have to increase the CSeq before
> >>forwarding the request, if the digit portion value has already
> >>been used in a request sent by the SBC in the same direction.
> >
> >That seems like a good opportunity, within that trust domain
> >between the SBC and the other entity, to use
> >P-Asserted-Identity.
> 
> I am not sure I understand. How could the P-A-I header be 
> used instead of the CSeq?

The SBC that wants to interfere with the CSeq when doing 'dialog
piggybacking' can go ahead and do that.  Doing that will destroy 
the validity of the SIP-Identity-Media signature for the 'dialog 
piggybacked' request.  Because the SBC and the 'other entity' 
(which is receiving the 'dialog piggybacked' message) are both
in the same trust domain, they can use P-A-I between themselves.

-d


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Subject: RE: [Sip] WGLC for draft-ietf-sip-sips-05.txt  (v.minor error)
Date: Mon, 16 Jul 2007 12:54:06 -0500
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Thanks.

I will correct.=20

> -----Original Message-----
> From: Attila Sipos [mailto:attila.sipos@vegastream.com]=20
> Sent: Tuesday, July 03, 2007 03:15
> To: sip@ietf.org
> Subject: RE: [Sip] WGLC for draft-ietf-sip-sips-05.txt (v.minor error)
>=20
>=20
>=20
> See flows sections 6.3 and 6.4.
>=20
> F13' shows an INVITE to Bob's PC but the INVITE message shows:
> INVITE sip:bob@bobphone.example.com
>=20
> it should be:
> INVITE sip:bob@bobpc.example.com
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Mon Jul 16 14:03:46 2007
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Subject: RE: [Sip] WGLC for draft-ietf-sip-sips-05.txt
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Thanks.=20

I will make all those changes.=20

> -----Original Message-----
> From: Hans Persson [mailto:hasse@ingate.com]=20
> Sent: Thursday, July 05, 2007 02:25
> To: Audet, Francois (SC100:3055)
> Cc: Dean Willis; IETF SIP
> Subject: Re: [Sip] WGLC for draft-ietf-sip-sips-05.txt
>=20
> fre 2007-06-29 klockan 09:42 -0500 skrev Dean Willis:
>=20
> > I'm pleased to announce that the authors think that the=20
> SIPS draft is=20
> > ready for working group last call. This draft has had the=20
> privilege of=20
> > more review prior to WGLC than most of our other efforts,=20
> so perhaps=20
> > it won't be too controversial.
> >=20
> > Please make a final read through the draft and bring any=20
> comments back=20
> > to this list.
>=20
>=20
> I've gone through the diff between the current version and=20
> the last one
> I read, and on the whole it looks good to me.=20
>=20
> Some editorial nits are listed below.
>=20
>=20
> Section 3.1.1, para 1:
>=20
> > the Proxy is the TLS server side.  This directionality implies that
>=20
> "proxy"
>=20
> > directions (e..g, an incoming call), the UA must keep the TLS
>=20
> "e.g."
>=20
>=20
> Section 3.1.1, para 2:
>=20
> > connection also solves the NAT and Firewall problem as it ensures
>=20
> "firewall"
>=20
>=20
> Section 3.1.2, para 1:
>=20
> > place, the UAC would therefore take on the role of the TLS=20
> Client and
>=20
> "client"
>=20
> > SIP, a UA or a Proxy act both as UAC and UAS depending on=20
> if they are
>=20
> "proxy"
>=20
> > is very convenient.  This allows for TLS connections to be set-up or
>=20
> "set up"
>=20
>=20
> Section 3.1.2, para 3:
>=20
> > most residential SIP deployments for example.  Mutual TLS=20
> can be used
>=20
> Insert comma after "deployments".
>=20
> > in those environments, but only if the connection is always=20
> stated by
>=20
> "stated" is a strange word here. Do you mean "started"?
>=20
>=20
> Section 3.1.2, last para:
>=20
> > For this reasons, mutual TLS is mostly used in server-to-server
>=20
> "this" -> "these"
>=20
>=20
> Section 3.1.3, para 4:
>=20
> > message (From, To, Request-URI, Contact header field, Route, etc.).
>=20
> Why "Contact header field" and not just "Contact" like for the others?
>=20
>=20
> Section 3.4, para 6:
>=20
> > allowed, which would preclude the use of NATs or Firewalls.
>=20
> "firewalls"
>=20
>=20
> Section 4.1, para 3:
>=20
> > SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed).  UAS that
> >   do not support the SIPS URI scheme altogether "SHOULD reject the
>=20
> "UAS" would be more readable as "UASs".
>=20
> Change "altogether" to "at all".
>=20
>=20
> Section 4.1, last para:
>=20
> > response code 419 (SIPS Required)
>=20
> Missing period at end of sentence.
>=20
>=20
> Section 4.1.2, para 8:
>=20
> > reject requests using the SIP Scheme with 419 (SIPS Required).  A UA
>=20
> "scheme"
>=20
>=20
> Section 4.2, para 5:
>=20
> > response, it MUST NOT recurse on it.  The Proxy SHOULD forward the
>=20
> "proxy"
>=20
>=20
> Section 4.3, para 1:
>=20
> > Using a Redirect Server with TLS instead of using a Proxy has some
>=20
> "redirect server", "proxy"
>=20
> > established connection between the Proxy and the UAS (such as with
>=20
> "proxy"
>=20
>=20
> Section A.3:
>=20
> > (similar to what the transport=3Dtls used to mean) has been =
suggested
>=20
> Remove "the".
>=20
>=20
> Hans
>=20
> --=20
> Hans Persson <hasse@ingate.com>    Ingate - Firewalls with SIP & NAT
> Ingate Systems AB  +46 13 210857   http://www.ingate.com/
>=20
> Private: <unicorn@lysator.liu.se>  http://www.lysator.liu.se/~unicorn/
>=20


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From sip-bounces@ietf.org Mon Jul 16 14:06:20 2007
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Subject: Re: [Sip] SIP Location Conveyance-08 is available
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Hi James,
	
>
>- added a Geolocation header parameter "recipient=" to indication which SIP entity type a particular locationValue is intended for.
>
>        OPEN ISSUE here: what does a Location Recipient
>                do if more than one locationValue point to it? (i.e.,
>                the case in which the UAS receives 2 locations
>                and both state "recipient=endpoint")

First a couple of nits, then a question on the open issue.

Section 3.2 says:
      the "recipient=" to allow recipients to infer what SIP element
      type this locationValue was intended to be for.  The types are
      "endpoint", meaning the ultimate destination UAS;
      "routing-entity", meaning SIP servers; and "both" meaning this
      locationValue is to be viewed by both types of SIP entities. 

The example in 4.2, though, has "recipient=server".  That should
get shifted to "routing-entity", if I understand things correctly.

The document also points to 3261 for Absolute URI, which appears to
derive that from RFC 2396, Section 5.  Would it be a problem to update
that to RFC 3986, Section 4.3, since 3986 is now a full standard?

So, the open issue you raise is what a routing-entity or endpoint
does when it receives multiple locationValues for which it is the
recipient.  For the endpoint, I believe that the answer is
application specific.  For some uses, the most recently added may
get used; for others, the first added; for others, all of them
may get evaluated and some other application-specific judgement call
made (an application might prefer the most precise, a specific
address type, or something else).  I think, though, that this
document can stay silent about the endpoint's selection beyond the error
code text describing 424.

For the routing-entity, though, I think that silence is probably
not workable.  I would personally say that a routing-entity receiving
multiple locationValues should always use the most recently
added unless that location would generate a 424; in that case, it
can examine the previously asserted locations in reverse order of
addition until it finds one that is valid or generates the 424.

Would that make sense to you?
				regards,
					Ted





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Subject: RE: [Sip] draft-ietf-sip-sips-05.txt - comment on response codes 418
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Thread-Topic: [Sip] draft-ietf-sip-sips-05.txt - comment on response codes 418
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From: "Francois Audet" <audet@nortel.com>
To: "Attila Sipos" <attila.sipos@vegastream.com>, <sip@ietf.org>
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My personal opinion is that those are good suggestions. But it does mean
2 new headers as well. I also like the idea of having only one response
code (i.e., 418 Invalid URI scheme).

Any other opinions?

I will flag this as a remaining issue in Chicago.

> -----Original Message-----
> From: Attila Sipos [mailto:attila.sipos@vegastream.com]=20
> Sent: Monday, July 09, 2007 00:51
> To: sip@ietf.org
> Subject: [Sip] draft-ietf-sip-sips-05.txt - comment on=20
> response codes 418 and 419
>=20
>=20
> This draft specifies two new response codes:
>=20
> 418 SIPS Not Allowed
> 419 SIPS Required
>=20
> With sipsec around the corner shouldn't we make these headers=20
> more generic like we do for other SIP things?
>=20
> For example, if someone uses a method we don't allow then we=20
> return "405 Method Not Allowed" and then list the allowed=20
> methods in the Allow header.
> And there is a similar solution for SIP extensions (420 Bad=20
> Extension or 421 Extension Required).
>=20
> So how about:
> 418 URI Scheme Not Allowed
> Allow-URI: sip
>=20
> 419 URI Scheme Required
> Require-URI: sips,sipsec
>=20
> In fact, you probably only need one of them as they are=20
> almost identical in resultant behaviour.
>=20
> Regards,
>=20
> Attila
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From: "Dan Wing" <dwing@cisco.com>
To: "'Christer Holmberg \(JO/LMF\)'" <christer.holmberg@ericsson.com>
Subject: RE: [Sip] SIP Identity using Media Path
Date: Mon, 16 Jul 2007 14:43:09 -0700
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I wrote:

> > >>The CSeq (at least the digit portion) may also be modified,
> > >>for example if there has been some "dialog piggybacked" requests
> > >>sent between the SBC and another entity, but not end-to-end. In
> > >>that case the the SBC may have to increase the CSeq before
> > >>forwarding the request, if the digit portion value has already
> > >>been used in a request sent by the SBC in the same direction.
> > >
> > >That seems like a good opportunity, within that trust domain
> > >between the SBC and the other entity, to use
> > >P-Asserted-Identity.
> > 
> > I am not sure I understand. How could the P-A-I header be 
> > used instead of the CSeq?
> 
> The SBC that wants to interfere with the CSeq when doing 'dialog
> piggybacking' can go ahead and do that.  Doing that will destroy 
> the validity of the SIP-Identity-Media signature for the 'dialog 
> piggybacked' request.  Because the SBC and the 'other entity' 
> (which is receiving the 'dialog piggybacked' message) are both
> in the same trust domain, they can use P-A-I between themselves.

I sketched out what you had said, and I now understand what you
were saying.  The case is like this:

              +----+
              |    +------CSeq=2--(dialog piggybacked request-->
  -CSeq=1---->+ SBC|
              |    +------CSeq=1--(request)-->
              +----+

What I had suggested was that the piggybacked request (the one
with CSeq=2) could be sent with P-A-I (created by the SBC), as 
it was being sent to another entity within the same trust domain
as the SBC.

The difficulty you were describing was that when the next request
arrives, it may be using CSeq=2:

              +----+
              |    +-->
  -CSeq=2---->+ SBC|
              |    +-->
              +----+

the SBC would need to disambiguate the newly-arrived CSeq from
the piggybacked (spoofed) request the SBC created from the
previous request.

The easy solution to this is to not sign the CSeq's sequence
number, but just the method.  For replay protection we would
rely on the nonce created by the signing service (which I
described in
http://www1.ietf.org/mail-archive/web/sip/current/msg19883.html).

I can't read the tea leaves of RFC4474 to understand why RFC4474
signs CSeq's sequence number.


Let me know if removing CSeq's sequence number from the signature
would resolve the problem you brought up.

-d


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Thread-Topic: ICE deployment data before LC for RFC
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
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The work on ICE is truly impressive and so are the numerous I-Ds
associated with ICE.

=20

However, before sending the <ice-17> I-D for LC to the IESG, it would be
prudent and responsible to the industry (that spends considerable
resources on ICE in good faith) implementing ICE, to either (1) make it
an informational RFC or (2) publish some deployment data showing such
items as:

=20

*	NAT scenarios that have been tested,
*	The % of success,
*	Performance, such as call setup delay using SIP.

=20

I have not seen any such public data on ICE deployment, for example with
SIP. A private report has raised my concern about the performance of
ICE.

=20

In the best IETF tradition, it would also help to have open source code
available for ICE, before declaring it a standard.

=20

ICE is too important for the IETF (good work Jonathan and other
authors!) not to publish (1) deployment results and (2) publish open
source code as well.

The various nits discussed on the lists are only of secondary importance
to the LC compared to the above and would get resolved with such a
process anyway.

=20

These are my personal two cents.

Henry Sinnreich




=20


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The work on ICE is truly impressive and so are the =
numerous
I-Ds associated with ICE.<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'>However, before sending the &lt;ice-17&gt; I-D for LC =
to the
IESG, it would be prudent and responsible to the industry (that spends
considerable resources on ICE in good faith) implementing ICE, to either =
(1)
make it an informational RFC or (2) publish some deployment data showing =
such
items as:<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>

<ul style=3D'margin-top:0in' type=3Ddisc>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo3'><font size=3D2 =
face=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>NAT scenarios that =
have been
     tested,<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo3'><font size=3D2 =
face=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The % of =
success,<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo3'><font size=3D2 =
face=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>Performance, such as =
call setup
     delay using SIP.<o:p></o:p></span></font></li>
</ul>

<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'>I have not seen any such public data on ICE =
deployment, for
example with SIP. A private report has raised my concern about the =
performance
of ICE.<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'>In the best IETF tradition, it would also help to =
have open
source code available for ICE, before declaring it a =
standard.<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'>ICE is too important for the IETF (good work Jonathan =
and
other authors!) not to publish (1) deployment results and (2) publish =
open
source code as well.<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'>The various nits discussed on the lists are only of
secondary importance to the LC compared to the above and would get =
resolved with
such a process anyway.<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'>These are my personal two =
cents.<o:p></o:p></span></font></p>

<p style=3D'margin-bottom:12.0pt'><st1:PersonName w:st=3D"on"><font =
size=3D2
 face=3D"Times New Roman"><span style=3D'font-size:10.0pt'>Henry =
Sinnreich</span></font></st1:PersonName><br>
<br>
<br>
<o:p></o:p></p>

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=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: Monday, July 09, 2007 05:55
> To: Audet, Francois (SC100:3055); sip@ietf.org
> Subject: WGLC comments on sip-sips
>=20
> In general this is now in excellent shape. Just some detailed=20
> comments.
>=20
> 1. "It also makes normative changes to SIP". If so, then it=20
> should state on the front page that it updates RFC 3261.

Agreed. Will change.

> 2. In 3.1.1 "In this model, only the TLS server provides a=20
> certificate during the
>    TLS handshake.  Often, a proxy (which may or may not be the one
>    terminating the TLS connection) authenticates the client=20
> by using the
>    SIP HTTP digest.  In this model, the UA is the TLS client side, and
>    the Proxy is the TLS server side."
> The words "the Proxy" in the third sentence would appear to=20
> refer to "a proxy" in the second sentence, but that doesn't=20
> make sense. Also I think this is trying to say that the=20
> server-provided certificate can only apply between a UA and a=20
> proxy and when the UA is the client, so it can't apply=20
> between proxies and it can't apply when a proxy is a client.
> If so, to solve these two issues, perhaps it should say.
> "In this model, only the TLS server provides a certificate=20
> during the TLS handshake.  This is applicable only between a=20
> UA and a proxy, where the UA is the TLS client and the proxy=20
> is the TLS server, and hence the UA uses TLS to authenticate=20
> the proxy but the proxy does not use TLS to authenticate the=20
> UA. If the proxy needs to authenticate the UA, this can be=20
> achieved by SIP HTTP digest authentication."

Yes, much clearer. I'll fix.

> 3. In 3.1.2 "In this model, both the TLS client and the TLS=20
> server provide a
>    certificate in the TLS handshake phase.  When no TLS=20
> connection is in
>    place, the UAC would therefore take on the role of the TLS=20
> Client and
>    the UAS would therefore take on the role of the TLS server."
> This seems to limit mutual TLS (more correctly TLS mutual
> authentication) to use involving a UA, but it also has an=20
> important use between proxies (this is not mentioned until=20
> further down). Also the second sentence seems to suggest a=20
> direct TLS connection between UAC and UAS, which is possible=20
> but not the typical case. Where at least one proxy is=20
> involved in the signalling path, if there is not already a=20
> TLS connection in place between the UAC and the first proxy,=20
> the UAC would take on the role of TLS client and the first=20
> proxy would take on the role of TLS server. Likewise, if=20
> there is not already a TLS connection in place between the=20
> last proxy and the UAS, the last proxy would take on the role=20
> of the TLS client and the UA would take on the role of the=20
> TLS server. I think this needs to be explained more fully.=20
> Perhaps we need to say something like:
> "In this model, both the TLS client and the TLS server=20
> provide a certificate in the TLS handshake phase. When used=20
> between a UA and a proxy (or between two UAs), this implies=20
> that a UA must be in possession of a certificate. When=20
> sending a SIP request when there is not already a suitable=20
> TLS connection in place, a UAC takes on the role of TLS=20
> client in establishing a new TLS connection. When=20
> establishing a TLS connection for receipt of a SIP request, a=20
> UAS takes on the role of TLS server."

Yes, much clearer. I'll fix (but I'll keep the last sentence of the
paragraph).

> 4. "While guessing a SIPS AOR from a known SIP
>    AOR and using it to initiate a request is a valid thing to=20
> do, doing
>    the opposite (i.e., guessing a SIP AOR from a SIPS AOR and=20
> using it)
>    is not a valid thing to do as it would be a security downgrade."
> I would not go as far as saying it is not valid. It is merely=20
> undesirable, but if the caller's UA does not support the SIPS=20
> scheme, that might be a valid way of contacting the desired user.

Ok, will change.

> 5. "Although "downgrading" from SIPS to SIP is disallowed,"
> As mentioned above, I think it is valid but undesirable for a=20
> user to downgrade by guessing. However, I think what we are=20
> talking about here is downgrading by a proxy, so perhaps that=20
> should be made clear by
> saying: "Although "downgrading" from SIPS to SIP is=20
> disallowed at a proxy,".

Ok, will change.

> 6. In 4.1 "If a UAS does not wish to be reached with a SIPS=20
> URI but only with a
>    SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed)."
> If the UA does not wish to be reached with a SIPS URI, then=20
> surely it would not register a SIPS contact. If the UA has=20
> not registered a SIPS contact, it should not normally receive=20
> a SIPS request. So presumably this is required only for=20
> requests reaching the UA without going through the domain=20
> proxy. Correct? If so, perhaps some explanation is needed.

The main use case is for endpoints that don't register (e.g.,
gateways). Furthermore, this is written from the UA point of view
and thus it could happen in error scenarios.=20

I'm not sure explaining this will help, but I can add it if=20
the groups feels it is necessary (I personally think it will just
make the text even more heavy).

> 7. "UAS that
>    do not support the SIPS URI scheme altogether "SHOULD reject the
>    request with a 416 (Unsupported URI scheme) response" as=20
> described in
>    [RFC3261]/8.2.2.1."
> Similar - should explain how this can occur.

Same as above.

> 8. In 4.1.1 "If a UA registers with a SIPS Contact header=20
> field, the registrar
>    returning a service route [RFC3608] MUST return a service route
>    consisting of SIP URIs if the intent of the registrar is to allow
>    both SIP and SIPS to be used in requests sent by that client.  If a
>    UA registers with a SIPS Contact header field, the registrar
>    returning a service route MUST return a service route consisting of
>    SIPS URIs if the intent of the registrar is to allow only SIPS URIs
>    to be used in requests sent by that UA."
> Because 4.1.1 is part of 4.1, which deals only with UAs, then=20
> it should not contain normative requirements on registrars.=20
> Presumably the intention of this is to state that RFC 3608=20
> places such a requirement on registrars. The language needs=20
> changing to reflect this.

Actually, Registrars are "a special type of UAS" according to RFC 3261
p. 56. This is why it's in this section.

> 9. "However, the registrar MUST treat the SIP and SIPS=20
> schemes of the AOR
>    the same way (i.e., it MUST not care if it is SIP or SIPS)."
> This is a requirement on the registrar and does not belong in 4.1.

Same as 8.

> 10. "A registrar MUST only accept a binding to a SIPS Contact=20
> header field
>    if all the appropriate URIs are of the SIPS scheme, otherwise there
>    could be an inadvertent binding of a secure resource (SIPS) to an
>    unsecured one (SIP).  This includes the Request-URI, the=20
> Contacts and
>    all the Path headers, but does not include the From and To headers.
>    If the URIs are not of the proper SIPS scheme, the registrar MUST
>    reject the REGISTER with a 419 (SIPS Required)."
> These are requirements on the registrar and do not belong in 4.1.

Same as 8.

> 11. In 4.2 "It is RECOMMENDED to use an outbound proxy as per=20
> the procedures
>    defined in [I-D.ietf-sip-outbound] for supporting UACs that can not
>    provide a certificate for establishing a TLS connection (i.e., when
>    server-side authentication is used)."
> This is a recommendation on a UA, not a proxy, so it does not=20
> belong as a normative statement in 4.2.

I'm not sure I agree on this. Seems like the decision to use or not the=20
procedures of sip-outbound in a proxy is a proxy requirement, not a UA=20
requirement.

> 12. In 4.3 "A
>    redirect server would not be usable if server-side=20
> authentication is
>    used"
> This is incorrect. If a UA has a certificate it can provide=20
> TLS server-side authentication, yet a redirect server could=20
> still be used in this case. I think this is trying to say=20
> that a redirect server cannot be used to redirect to an=20
> entity (e.g., a UA) that does not have a certificate.

Correct. I'll fix.

> 13. "When a redirect server receives a request with a SIP=20
> Request-URI, the
>    redirect server MAY redirect with a 3XX response to either=20
> a SIP or a
>    SIPS Contact header field.  If the target UAS had registered
>    previously using a SIPS Contact header field, the redirect server
>    SHOULD return a SIPS Contact header field if it is in an=20
> environment
>    where TLS is usable (as described in the previous paragraph)."
> This assumes that a redirect server is redirecting directly=20
> to a UA, i.e., to a contact. However, a redirect server can=20
> also redirect to another AoR that resolves to a domain proxy=20
> (or another redirect server). In other words, the redirect=20
> server may not have any knowledge of a "target UAS".
>=20
> 14. In 5.2 "Furthermore, the offending SIPS URI(s) may be in any
>    location in the request (e.g., the Request-URI, a Contact header
>    field, a Refer-To URI, etc.)."
> Shouldn't this say "the offending SIP URI(s)"?

Yes, I'll fix.

> 15. In the example in 6.3, proxy A and proxy B violate the=20
> recommendation in 4.2 to route SIP requests over TLS. This=20
> doesn't seem to be a very good example! It would be better to=20
> show routing over TLS from proxy A onwards.

I'll put some wording around this (i.e., go look at 6.4).

> 16. "7.  Conclusion"
> Most of the content of this section consists of further=20
> considerations, rather than conclusions drawn from previous=20
> sections. So perhaps the title should be "Further considerations".

Ok. I'll change it.

> 17. NITS
>=20
> - In 6.1 "This call flow". There is no call flow - the flow=20
> shows only registration.
>=20
> - There are instances of a SIP method name (e.g. REGISTER)=20
> being used alone without being followed by "request".
>=20
> - There are instances of "header" being used instead of=20
> "header field".

I'll fix the nits.

> John
>=20
>=20
>=20
>=20
>=20


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Subject: [Sip] ICE tutorial in Chicago [DO NOT REPLY] - PLEASE ADVISE IF YOU
 WILL ATTEND!
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I've gotten very few responses indicating interest. If I don't get more 
acceptances I'll have to cancel. Please, if you do plan on attending, 
send me mail per the instructions below if you have not done so.


(Apologies to multiple recipients)
(DO NOT REPLY)

WHAT:     Tutorial on Interactive Connectivity Establishment (ICE)
       (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-16.txt)
           [-17 should be available shortly]

WHEN:     Tuesday, July 24, 2007, 1145-1245
           (lunch logistics are in progress and will be announced ASAP)

WHERE:    Red Lacquar

WHO:      Anyone that would like to learn more about ICE.

WHY:      ICE, developed by the mmusic working group, has been
           defined as the IETF tool for NAT traversal for SIP-based
           media. It is also seeing increased usage for traversal
           for other kinds of application traffic - anything worrying
           about any-to-any communications. In this tutorial, we will
           overview the background of the problem, consider the
           requirements driving ICE, and provide an overview of
           how ICE works. An understanding of SIP is not required.
           The tutorial will be based on the pending -17 version.

RSVP:     Please send a note to me with the Subject line "ice-is-nice"
           (mailto:jdrosen@cisco.com?Subject=ice-is-nice) so I can get a
           count of the number of participants for lunch purposes.


!DO NOT REPLY TO THIS NOTE!
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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International Conference on IP Multimedia Subsystems Architecture and
Applications-2007 (IMSAA-2007),

http://www.iiitb.ac.in/imsaa2007/index.html ,

6th-8th December 2007, Bangalore, India,

by IEEE Bangalore-Section and International Institute of Information
Technology =96 Bangalore (iiit-b).

*Call for Papers*: http://www.iiitb.ac.in/imsaa2007/callforpapers.html



*Important Dates:*

*Last Date for Registration of Paper (Title and Abstract): *31st July, 2007
*Last Date for* *Complete Paper Submission*:  6th August, 2007
Paper review reports: 15th September 2007
Submission of the Camera ready papers: 5th October 2007

The IMSAA 2007 technical program committee is soliciting papers
addressing the research challenges and advances towards a world of
converged and intelligent services, based on novel applications and
extensions to the IMS framework. Papers must present original and
previously unpublished work, validated by experimentation, simulation,
or analysis. Practical experiences and experimental efforts, including
submissions from industry, including both telecom providers and device
vendors, addressing both successes and limitations with IMS, are
particularly welcome. Topics of interest include, but are not limited to
the following:



   - Architectures and Runtimes for IMS Elements


   - Application Server Architectures and Converged Containers
      - Architectures for Rich and Scalable Presence
      - Service Mediation and Billing Platforms


   - IMS Core Services


   - Real-time Service Support: (voice, video calls, multimedia
      conferencing, multiplayer gaming)
      - Content/Context-aware services: (Location, presence, social
      and community context)
      - Service Creation Environments (IMS, SCIM, Service Delivery
      Platforms, Integrated environments)


   - Device,  Subscriber, Service and Network Management


   - Provisioning and management (Handsets, PDA's, Set top Box, Home
      Gateway)


   - Subscriber and User Profile Management
      - Managing  the quality of the user experience (QoE, QoS)
      - End to end Performance Measurement  and modeling  (signaling,
      media, services)


   - Operator Network Planning


   - Routing and Load Balancing


   - Redundancy and Fault Tolerance
      - MVNO Operations and Support


   - Novel IMS-based Applications


   - Gaming and mixed-reality
      - Environmental and Sensor Monitoring
      - Multimedia Content Sharing
      - Converged IN
      - Converged Home Network
      - IPTV over IMS


   - Security, Identity  Management and Privacy


   - Privacy and Security in Composite Applications
      - Privacy in LBS


   - IMS and Alternative Technologies


   - P2P and IMS (VoIP, Instant Messaging, Gaming)
      - Integrating IMS and Web2.0 services  (Mashups, Rich Internet
      Applications)
      - User-generated Services and Content
   - Case Studies and Status of IMS deployment


   - Interoperability Testing
      - IMS Support in Vendor Products


   - Benchmarking IMS Services and Applications


   - Testbeds and Software Tools


   - Service Test and Validation Platforms
      - Open Source Initiatives
      - Benchmarking IMS Services and Applications


   - IMS Clients and Devices


   - IMS-aware consumer devices
      - Media/communication platform s

*Paper Submission Format:*
Papers must be submitted in IEEE format (.pdf file). The EDAS Website
for uploading the paper is http://edas.info/newPaper.php?c=3D5595.  Papers
for review: Double-column, 11pt IEEE format and no longer than 7 pages
(including all figures and references). Camera ready paper: It will be
in double-column, IEEE format, but in 10 pt font, and no longer than 6
pages (including all figures and references).

 Technical Committee: http://www.iiitb.ac.in/imsaa2007/technicalcomm.html
-----------------------------------------------------------------------

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<p class=3D"MsoNormal">&nbsp;</p>

<p class=3D"MsoNormal">International Conference on IP Multimedia Subsystems
Architecture and Applications-2007 (IMSAA-2007), <br></p><p class=3D"MsoNor=
mal"><a href=3D"http://www.iiitb.ac.in/imsaa2007/index.html">http://www.iii=
tb.ac.in/imsaa2007/index.html</a>
, <br></p><p class=3D"MsoNormal">6th-8th December 2007, Bangalore, India, <=
br></p><p class=3D"MsoNormal">by IEEE Bangalore-Section and
International Institute of Information Technology =96 Bangalore (iiit-b).<b=
r>
<br>
<b><i>Call for Papers</i></b>: <a href=3D"http://www.iiitb.ac.in/imsaa2007/=
callforpapers.html">http://www.iiitb.ac.in/imsaa2007/callforpapers.html</a>=
</p>

<p class=3D"MsoNormal">&nbsp;</p>

<p class=3D"MsoNormal"><a name=3D"OLE_LINK1"><b>Important Dates:</b></a></p=
>

<p class=3D"MsoNormal"><span style=3D""><b><span style=3D"color: red;">Last=
 Date for Registration of Paper (Title and Abstract): </span></b><span styl=
e=3D"color: red;">31<sup>st</sup> July, 2007<br>
<b>Last Date for</b> <b>Complete Paper Submission</b>:<span style=3D"">&nbs=
p; </span>6<sup>th</sup> August, 2007</span><br>
</span>Paper review reports: 15th September 2007<br>
Submission of the Camera ready papers: 5th October 2007<br>
<br>
The IMSAA 2007 technical program committee is soliciting papers <br>
addressing the research challenges and advances towards a world of <br>
converged and intelligent services, based on novel applications and <br>
extensions to the IMS framework. Papers must present original and <br>
previously unpublished work, validated by experimentation, simulation, <br>
or analysis. Practical experiences and experimental efforts, including <br>
submissions from industry, including both telecom providers and device <br>
vendors, addressing both successes and limitations with IMS, are <br>
particularly welcome. Topics of interest include, but are not limited to <b=
r>
the following:<br style=3D"">
<br style=3D"">
</p><br><ul><li>Architectures and Runtimes for IMS Elements </li></ul>
      <ul><ul><li>Application Server Architectures and Converged Containers=
&nbsp;
          </li><li>Architectures for Rich and Scalable Presence=20
          </li><li>Service Mediation and Billing Platforms </li></ul></ul>
      <ul><li>IMS Core Services </li></ul>
      <ul><ul><li>Real-time Service Support: (voice,&nbsp;video calls, mult=
imedia conferencing, multiplayer gaming)
          </li><li>Content/Context-aware services:&nbsp;(Location, presence=
, social and community context)
          </li><li>Service Creation Environments (IMS, SCIM, Service Delive=
ry Platforms, Integrated environments) </li></ul></ul>
      <ul><li>Device,&nbsp; Subscriber, Service and Network Management </li=
></ul>
      <ul><ul><li>Provisioning and management (Handsets, PDA&#39;s, Set top=
 Box, Home Gateway) </li></ul></ul>
      <ul><ul><li>Subscriber and User Profile Management=20
          </li><li>Managing&nbsp; the quality of the user experience (QoE, =
QoS)
          </li><li>End to end Performance Measurement&nbsp; and modeling&nb=
sp; (signaling, media, services) </li></ul></ul>
      <ul><li>Operator Network Planning&nbsp; </li></ul>
      <ul><ul><li>Routing and Load Balancing </li></ul></ul>
      <ul><ul><li>Redundancy and Fault Tolerance=20
          </li><li>MVNO Operations and Support </li></ul></ul>
      <ul><li>Novel IMS-based Applications </li></ul>
      <ul><ul><li>Gaming and mixed-reality=20
          </li><li>Environmental and Sensor Monitoring=20
          </li><li>Multimedia Content Sharing=20
          </li><li>Converged IN=20
          </li><li>Converged Home Network=20
          </li><li>IPTV over IMS </li></ul></ul>
      <ul><li>Security, Identity&nbsp; Management and Privacy&nbsp; </li></=
ul>
      <ul><ul><li>Privacy and Security in Composite Applications=20
          </li><li>Privacy in LBS </li></ul></ul>
      <ul><li>IMS and Alternative Technologies </li></ul>
      <ul><ul><li>P2P and IMS (VoIP, Instant Messaging, Gaming)=20
          </li><li>Integrating IMS and Web2.0 services&nbsp; (Mashups, Rich=
 Internet Applications)
          </li><li>User-generated Services and Content </li></ul><li>Case S=
tudies and Status of IMS deployment </li></ul>
      <ul><ul><li>Interoperability Testing=20
          </li><li>IMS Support in Vendor Products </li></ul></ul>
      <ul><ul><li>Benchmarking IMS Services and Applications </li></ul></ul=
>
      <ul><li>Testbeds and Software Tools </li></ul>
      <ul><ul><li>Service Test and Validation Platforms=20
          </li><li>Open Source Initiatives=20
          </li><li>Benchmarking IMS Services and Applications </li></ul></u=
l>
      <ul><li>IMS Clients and Devices </li></ul>
      <ul><ul><li>IMS-aware consumer devices=20
          </li><li>Media/communication platform s </li></ul></ul><p class=
=3D"MsoNormal">
<b>Paper Submission Format:</b><br>
Papers must be submitted in IEEE format (.pdf file). The EDAS Website <br>
for uploading the paper is <a href=3D"http://edas.info/newPaper.php?c=3D559=
5">http://edas.info/newPaper.php?c=3D5595</a>.&nbsp;
Papers <br>
for review: Double-column, 11pt IEEE format and no longer than 7 pages <br>
(including all figures and references). Camera ready paper: It will be <br>
in double-column, IEEE format, but in 10 pt font, and no longer than 6 <br>
pages (including all figures and references).</p>

<p class=3D"MsoNormal"><span style=3D"">&nbsp;</span>Technical Committee: <=
a href=3D"http://www.iiitb.ac.in/imsaa2007/technicalcomm.html">http://www.i=
iitb.ac.in/imsaa2007/technicalcomm.html</a><br>
-----------------------------------------------------------------------</p>


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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0117609741==--





From sip-bounces@ietf.org Tue Jul 17 02:07:28 2007
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Subject: RE: [Sip] SIP Identity using Media Path
Date: Tue, 17 Jul 2007 08:06:10 +0200
Message-ID: <7374777208BDC7449D5620EF942325670730B11C@esealmw113.eemea.ericsson.se>
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Thread-Topic: [Sip] SIP Identity using Media Path
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Dan Wing" <dwing@cisco.com>
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Hi Dan,

I was planning to provide an example myself, but you seem to have done
the work for me :)

>>>>>The CSeq (at least the digit portion) may also be modified, for
example if there has been some "dialog piggybacked"=20
>>>>>requests sent between the SBC and another entity, but not
end-to-end. In that case the the SBC may have to increase the CSeq
before=20
>>>>>forwarding the request, if the digit portion value has already been
used in a request sent by the SBC in the same direction.
>>>>
>>>>That seems like a good opportunity, within that trust domain=20
>>>>between the SBC and the other entity, to use P-Asserted-Identity.
>>>=20
>>>I am not sure I understand. How could the P-A-I header be used
instead of the CSeq?
>>=20
>>The SBC that wants to interfere with the CSeq when doing 'dialog=20
>>piggybacking' can go ahead and do that.  Doing that will destroy the=20
>>validity of the SIP-Identity-Media signature for the 'dialog=20
>>piggybacked' request.  Because the SBC and the 'other entity'
>>(which is receiving the 'dialog piggybacked' message) are both in the=20
>>same trust domain, they can use P-A-I between themselves.
>=20
>I sketched out what you had said, and I now understand what=20
>you were saying.  The case is like this:
>=20
>               +----+
>               |    +------CSeq=3D2--(dialog piggybacked request-->
>   -CSeq=3D1---->+ SBC|
>               |    +------CSeq=3D1--(request)-->
>               +----+
>=20
>What I had suggested was that the piggybacked request (the=20
>one with CSeq=3D2) could be sent with P-A-I (created by the=20
>SBC), as it was being sent to another entity within the same=20
>trust domain as the SBC.
>=20
>The difficulty you were describing was that when the next=20
>request arrives, it may be using CSeq=3D2:
>=20
>               +----+
>               |    +-->
>   -CSeq=3D2---->+ SBC|
>               |    +-->
>               +----+
>=20
>the SBC would need to disambiguate the newly-arrived CSeq=20
>from the piggybacked (spoofed) request the SBC created from=20
>the previous request.

Yes. The SBC would need for forward the request with an increased CSeq
value, e.g. Cseq=3D3.

Having said that, I think it's quite unlikely that the SBC would
piggyback anything before it receives the initial INVITE, so for that
request it is maybe possible to assume that the CSeq will stay
unchanged.

But, during the dialog the SBC may have to change the Cseq digit value,
and I've also seen use-cases where the method is changed.

Regards,

Christer


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



From sip-bounces@ietf.org Tue Jul 17 02:34:35 2007
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Date: Tue, 17 Jul 2007 09:34:15 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Qian Sun <sunqian@huawei.com>
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	Aki.Niemi@nokia.com, Markus.Isomaki@nokia.com
Subject: [Sip] Re: [Sipping] RE: Comments on
	draft-ietf-sip-multiple-refer-01.txt
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Hi,

[moving this thread from the SIPPING list to this one (SIP) because this=20
is a SIP document now.]

You are right. It is not trivial to use Referred-By with multiple REFER.=20
My suggestion is that, the same way as Referred-By is specified as an=20
extension of refer, you go ahead specify (if you are interested) how to=20
use Referred-by with multiple REFER in a different spec.

The body that the referrer would add to its REFER (which is what the=20
refer target would get) could be an XML list with the recipients (those=20
that can be disclosed) of the multiple REFER (plus the referrer).=20
Something similar to what recipients get after a relay processes a list=20
with copyControl attributes=20
(draft-ietf-sipping-capacity-attribute-04.txt)... in any case, since=20
multiple REFER is only used within a context of a service, you need to=20
think whether having a general Referred-By mechanism is needed or if=20
such functionality can be achieved in a different way within the=20
services multiple REFER is used in (e.g., conferencing).

Cheers,

Gonzalo


Qian Sun wrote:
> Hello, Gonzalo
>=20
> RFC3892 says:
>    The sipfrag inside a Referred-By token MUST contain copies of the
>    Refer-To, Referred-By, and Date header fields from the REFER request=
.
>=20
> In multiple refer case, the Refer-To header field includes content id,=20
> not the REFER-Target' URI. The content id is meaningless to the=20
> REFER-Targets, so the Referred-By token has to include the URI-List.
>=20
> Cheers,
> Qian
>=20
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: Monday, July 16, 2007 4:23 PM
> To: Qian Sun
> Cc: Aki.Niemi@nokia.com; Markus.Isomaki@nokia.com;=20
> miguel.an.garcia@nokia.com; hisham.khartabil@gmail.com; sipping@ietf.or=
g
> Subject: Re: Comments on draft-ietf-sip-multiple-refer-01.txt
>=20
> Hi,
>=20
> thanks for your comments. Answers inline:
>=20
> Qian Sun wrote:
>  > Some comments on the draft...
>  >
>  > 1. I think it might be a good idea to talk something on "Referred-By=
"=20
> in this draft. Referred-By mechanism is still very useful for the=20
> multiple refer case. Maybe Refer-To related to a URI List with "bcc" or=
=20
> "anonymize" is a trouble. "Referred-BY=E2=80=9D in the message header i=
s to=20
> identify the initial REFER-Issuer. This information may be encrypted=20
> together with "refer-to" URI list, including the copy control level suc=
h=20
> as to/cc/bcc to the end user, i.e. REFER-Target, where the information=20
> will be decrypted by the REFER-Target. In this case the REFER-Recipient=
=20
> won=E2=80=99t be able to get rid of the "bcc" URI from the URI list, an=
d the=20
> "bcc" URI will be sent to other REFER-Targets along with normal to/cc.=20
> This will be a problem to destroy the blind cc function.
>  > However, the REFER-issuer may opt to not include a Referred-By token=
.=20
> For example:
>  >    REFER sip:conf-123@example.com;gruu;opaque=3Dhha9s8d-999a  SIP/2.=
0
>  >    Refer-To: <cid:cn35t8jf02@example.com>
>  >    Referred-By: Carol <sip:carol@chicago.example.com>
>  >    Content-Type: application/resource-lists+xml
>  >    Content-Disposition: recipient-list
>  >    Content-Length: 362
>  >    Content-ID: <cn35t8jf02@example.com>
>  >
>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>  >      <list>
>  >        <entry uri=3D"sip:bill@example.com?method=3DBYE" />
>  >        <entry uri=3D"sip:joe@example.org?method=3DBYE" />
>  >        <entry uri=3D"sip:ted@example.net?method=3DBYE" />
>  >      </list>
>  >    </resource-lists>
>  >
>  > When the URI list has no "bcc" or "anonymize" URI, REFER-issuer may=20
> include the Referred-By token in the REFER body.
>=20
> The way Referred-By works is not affected by what this draft says.
> Therefore, I do not see the need to talk about it.
>=20
>  >
>  > 2. Nested refer in multiple refer is very interesting, I believe it=20
> could be a good idea to enroll a "multiple nested refer" example into=20
> this draft. Just for an example:
>  >    REFER sip:list-service@example.com;gruu;opaque=3Dhha9s8d-999a  SI=
P/2.0
>  >    Refer-To: <cid:cn35t8jf02@example.com>
>  >    Content-ID: <cn35t8jf02@example.com>
>  >
>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>  >      <list>
>  >        <entry=20
> uri=3D"sip:bill@example.com?method=3DREFER%3FRefer-To%3Dsip:conf-123%40=
shenzhen.example.com"=20
> />
>  >        <entry=20
> uri=3D"sip:tom@example.com?method=3DREFER%3FRefer-To%3Dsip:conf-123%40s=
henzhen.example.com"=20
> />
>  >      </list>
>  >    </resource-lists>
>  > In this case, the REFER-Recipient acts as a Refer List service, has=20
> the similar function to the MESSAGE List service, i.e. receives a REFER=
=20
> with URI-List and distributes some REFERs to each address in the URI-Li=
st.
>  > Note that, there is only one final REFER-Target in above "nested=20
> multiple refer" example, not multiple REFER-Targets. In some situations=
,=20
> this convergent feature is useful.
>=20
> The draft says the following:
>=20
>     Additionally, REFER-Recipients SHOULD only accept REFER requests
>     within the context of an application the server understands (e.g., =
a
>     conferencing application).
>=20
> this text was added to avoid relays being used as random traffic
> forwarders. Therefore, since a relay needs to understand what is going
> on (e.g., a multiple REFER to trigger BYEs towards several conference
> attendees) and currently there is no service that uses nested refers, I
> do not think we need to talk about it.
>=20
>  >
>  > 3. I guess it might be a typo in Figure 2 "8. 200K OK", maybe you=20
> meant "200 OK"?
>=20
> yes, thanks for pointing it out. We will fix it.
>=20
> Thanks,
>=20
> Gonzalo
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20



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This list is for NEW development of the core SIP Protocol
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Date: Mon, 16 Jul 2007 23:42:21 -0700
From: "Nasir Khan" <nkhan.sip@gmail.com>
To: "Robert Sparks" <rjsparks@estacado.net>
Subject: Re: [Sip] Changes to the INVITE transaction to address bug 769
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Hi Robert,

I have 3 comments / suggestions w.r.t this ID.

Comment-1
----------------
Should there be any change with respect to receiving CANCEL in Accepted
state? Hitherto the UAC was getting a 481/CANCEL in such cases. Now we will
get 200/CANCEL with no effect on IST.
The reason for this question is backwards compatibility, however I can
myself argue both ways on this one. What do you think?


Comment-2
-----------------
In section 6.7 of the ID about the Timer-L you say - "and the amount of time
this (or any downstream) UAS core might be retransmitting the 2xx while
waiting for an ACK. "

How does 2xx retransmission timer affect the choice of Timer L?


Comment-3
-----------------
Shouldnt the Timer L be set to T4 which represents the amount of time the
network will take to clear messages between client and server transactions?
Since 2xx is sent reliably anyway, IMHO we do not need to link Timer L to
Timer B.

Thanks

Nasir Khan
BEA Systems Inc.


On 6/27/07, Robert Sparks <rjsparks@estacado.net> wrote:
>
> Everyone -
>
> I've put together a proposal for addressing the spec bug captured at
> http://bugs.sipit.net/show_bug.cgi?id=769
>
> Please read through
> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
> and comment.
>
> RjS
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>



-- 
Discuss SIP Servlets at http://groups.google.com/group/sipservlets/

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Hi Robert,<br>

<br>

I have 3 comments / suggestions w.r.t this ID. <br>

<br>

Comment-1<br>

----------------<br>

Should there be any change with respect to receiving CANCEL in Accepted
state? Hitherto the UAC was getting a 481/CANCEL in such cases. Now we
will get 200/CANCEL with no effect on IST. <br>

The reason for this question is backwards compatibility, however I can myself argue both ways on this one. What do you think?<br>

<br>

<br>

Comment-2<br>

-----------------<br>

In section 6.7 of the ID about the Timer-L you say - &quot;and the amount of
time this (or any downstream) UAS core might be retransmitting the 2xx
while waiting for an ACK. &quot; <br>

<br>

How does 2xx retransmission timer affect the choice of Timer L? <br>

<br>

<br>

Comment-3<br>

-----------------<br>

Shouldnt the Timer L be set to T4 which represents the amount of time
the network will take to clear messages between client and server
transactions? <br>

Since 2xx is sent reliably anyway, IMHO we do not need to link Timer L to Timer B.<br>

<br>

Thanks<br>

<br>

Nasir Khan<br>

BEA Systems Inc.<br>
<br><br><div><span class="gmail_quote">On 6/27/07, <b class="gmail_sendername">Robert Sparks</b> &lt;<a href="mailto:rjsparks@estacado.net">rjsparks@estacado.net</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Everyone -<br><br>I&#39;ve put together a proposal for addressing the spec bug captured at<br><a href="http://bugs.sipit.net/show_bug.cgi?id=769">http://bugs.sipit.net/show_bug.cgi?id=769</a><br><br>Please read through<br>
<a href="http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt">http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt</a><br>and comment.<br><br>RjS<br><br><br>_______________________________________________
<br>Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a href="mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip<br>Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br></blockquote></div><br><br clear="all"><br>
-- <br>Discuss SIP Servlets at <a href="http://groups.google.com/group/sipservlets/">http://groups.google.com/group/sipservlets/</a>

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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
--===============0830415077==--





From sip-bounces@ietf.org Tue Jul 17 03:55:00 2007
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi Linyi,

> (1) On section "5.  Conference Server Procedures"
> Many conference servers have restriction to maximum number of attendees 
> for conference. If the users of 'recipient-list' exceeds the limitation, 
> what status code should be returned?

I would return a 403 (Too many recipients).

> ---------------------------------------
> 7.  Security Considerations
>    This document discusses setup of SIP conferences using a 
> request-contained URI-list.  Both conferencing and URI-lists services 
> have specific security requirements which will be summarized here. 
> Conferences generally have authorization rules about who can or cannot 
> join a conference, what type of media can or cannot be used, etc.  This 
> information is used by the focus to admit or deny participation in a 
> conference.  It is RECOMMENDED that these types of authorization rules 
> be used to provide security for a SIP conference.
> ----------------------------------------
> (2) When and how the authorization rules are specified and applied to 
> the conference? Before the conference creation seems not possible since 
> Conf-URI is not assigned. If specified after the conference creation, it 
> seems there is a risk after conf is created but rules are not ready. 
> Some attendees may do something bad which is out of control during that 
> period.

They are applied at conference creation. The server receives the 
request, creates the conference with its rules, and only adds the 
participants in the list if those rules allow it to do it. This is 
already implemented in the OMA PoC (Push-to-talk over Cellular) service.

Thanks for your comments,

Gonzalo


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Francois,

More below. I have stripped out the comments that have been
satisfactorily dealt with.

John 

> 
> > 6. In 4.1 "If a UAS does not wish to be reached with a SIPS 
> > URI but only with a
> >    SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed)."
> > If the UA does not wish to be reached with a SIPS URI, then 
> > surely it would not register a SIPS contact. If the UA has 
> > not registered a SIPS contact, it should not normally receive 
> > a SIPS request. So presumably this is required only for 
> > requests reaching the UA without going through the domain 
> > proxy. Correct? If so, perhaps some explanation is needed.
> 
> The main use case is for endpoints that don't register (e.g.,
> gateways). Furthermore, this is written from the UA point of view
> and thus it could happen in error scenarios. 
> 
> I'm not sure explaining this will help, but I can add it if 
> the groups feels it is necessary (I personally think it will just
> make the text even more heavy).
[JRE] OK, I can live with that.

> 
> > 7. "UAS that
> >    do not support the SIPS URI scheme altogether "SHOULD reject the
> >    request with a 416 (Unsupported URI scheme) response" as 
> > described in
> >    [RFC3261]/8.2.2.1."
> > Similar - should explain how this can occur.
> 
> Same as above.
[JRE] Ditto.

> 
> > 8. In 4.1.1 "If a UA registers with a SIPS Contact header 
> > field, the registrar
> >    returning a service route [RFC3608] MUST return a service route
> >    consisting of SIP URIs if the intent of the registrar is to allow
> >    both SIP and SIPS to be used in requests sent by that 
> client.  If a
> >    UA registers with a SIPS Contact header field, the registrar
> >    returning a service route MUST return a service route 
> consisting of
> >    SIPS URIs if the intent of the registrar is to allow 
> only SIPS URIs
> >    to be used in requests sent by that UA."
> > Because 4.1.1 is part of 4.1, which deals only with UAs, then 
> > it should not contain normative requirements on registrars. 
> > Presumably the intention of this is to state that RFC 3608 
> > places such a requirement on registrars. The language needs 
> > changing to reflect this.
> 
> Actually, Registrars are "a special type of UAS" according to RFC 3261
> p. 56. This is why it's in this section.
[JRE] OK, I don't like it, but I suppose you are correct. However, you
didn't respond to the second part of my comment, concerning the need for
a change of language. Thinking about this further, I now see that we are
in fact updating RFC 3608 - correct? If so, then don't we need to state
this up front, and also mention it in the Introduction section where we
mention RFC 3261?

> 
> > 11. In 4.2 "It is RECOMMENDED to use an outbound proxy as per 
> > the procedures
> >    defined in [I-D.ietf-sip-outbound] for supporting UACs 
> that can not
> >    provide a certificate for establishing a TLS connection 
> (i.e., when
> >    server-side authentication is used)."
> > This is a recommendation on a UA, not a proxy, so it does not 
> > belong as a normative statement in 4.2.
> 
> I'm not sure I agree on this. Seems like the decision to use 
> or not the 
> procedures of sip-outbound in a proxy is a proxy requirement, 
> not a UA 
> requirement.
[JRE] The words "It is RECOMMENDED to use an outbound proxy" sound to me
like a recommendation on a UA, i.e., a recommendation that the UA be
configured to use an outbound proxy. Perhaps what we are trying to say
is "It is RECOMMENDED that a proxy be able to behave as an outbound
proxy as per the procedures...", in which case it would indeed belong in
this section.

> 
> > 13. "When a redirect server receives a request with a SIP 
> > Request-URI, the
> >    redirect server MAY redirect with a 3XX response to either 
> > a SIP or a
> >    SIPS Contact header field.  If the target UAS had registered
> >    previously using a SIPS Contact header field, the redirect server
> >    SHOULD return a SIPS Contact header field if it is in an 
> > environment
> >    where TLS is usable (as described in the previous paragraph)."
> > This assumes that a redirect server is redirecting directly 
> > to a UA, i.e., to a contact. However, a redirect server can 
> > also redirect to another AoR that resolves to a domain proxy 
> > (or another redirect server). In other words, the redirect 
> > server may not have any knowledge of a "target UAS".
[JRE] You didn't answer this or make any changes. Do you consider that
no change is necessary?



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From sip-bounces@ietf.org Tue Jul 17 05:53:06 2007
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To: ext Qian Sun <sunqian@huawei.com>
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Hi Qian,

And sorry for a late answer to your comment. I have been on vacation
enjoying the beautiful Finnish summer. ;)

ext Qian Sun wrote:
> Hi, I am not sure whether the following description in this draft is
> enough for the usage of Entity-tags with filtering.
> 
> "5.1.  Generating Entity-tags The views might correspond to different
> groups of users that have varying levels of access rights to the
> resource state, or to subscribers that have modified their
> subscription using event notification filtering [11]."
> 
> My consideration is: If a subscriber has a subscription with filter
> F1, the current corresponding entity-tag is E1. Then the same
> subscriber MUST NOT send SUBSCRIBE with a new filter F2 and the
> entity-tag E1 in the conditional header field. i.e., if a subscriber
> changes filter for an event state, the Suppress-XXX-If-Match should
> not be used. The client of subscriber has to record the last filter
> for event state and compare filters.

If I understand your concern correctly, the problem is with filtering
and not with subnot-etags. I think a subscription that installs a
filter, or disables/removes it could effectively be an action that
itself changes the event state, and causes an if-match condition to fail.

But this all depends on the actual implementation of the filtering
system. In other words, this seems like an implementation detail of the
filtering system.

At the moment, I am not aware of any existing filtering implementations,
so I'm reluctant to adding any specific text to the draft either. If
this becomes a problem, we can always write an update to the filtering
specifications.

Cheers,
Aki


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From sip-bounces@ietf.org Tue Jul 17 06:23:44 2007
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(As WG chair)

We reached the end of the time when WGLC comments were requested on
Monday.

If there are any last comments sitting out there that you have been
meaning to submit, can I suggest you get them in an quickly as possible.

Regards

Keith


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From sip-bounces@ietf.org Tue Jul 17 06:31:27 2007
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Hi Qian,

ext Qian Sun wrote:
> Hi,
> 
> As this draft mentioned, the presence rules may affect the views of
> the resource state. If the presence rules of presentity are modified,
> all of etags for this presentity have to be abandoned. Or the PS has
> to evaluate whether each etag still is valid.

I'm not sure I understand what it is that should be added to the draft
to clarify this issue.

There are all sorts of events that drive the internal state machine of
the PS, or more generally the server of any event package. For instance,
authorization rules might change, the resource state might change, and
those changes might or might not be visible to the subscriber. However,
those details are of no concern of the subnot-etags draft, IMO.

All that matters is that if the resource to which the subscriber
subscribed changes, then the entity-tag changes as well. What "change"
means here should be an implementation detail for every event package to
figure out. IOW, I don't think it belongs in the subnot-etags draft.

> For filtering, there is similar problem:
> http://www1.ietf.org/mail-archive/web/sip/current/msg19724.html
>
> These are difference between HTTP and SIP events framework.

Even HTTP resources can be dynamically generated, their access
controlled using an authorization framework of considerable complexity
and have many different views for different agents that access them.

So I don't think HTTP resources and SIP resources are that much
different in this respect.

Cheers,
Aki


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From sip-bounces@ietf.org Tue Jul 17 06:55:13 2007
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Date: Tue, 17 Jul 2007 11:54:23 +0100
From: "Elwell, John" <john.elwell@siemens.com>
To: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>, sip@ietf.org
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Mayumi,

"If the "temp-gruu" URI parameter and value exist within the REGISTER
   response, the user agent SHOULD use the value of the "temp-gruu" as
   an anonymous URI representing the originator.  This URI SHOULD be
   used for Contact and From, for example, wherever the originator of
   the URI is required."
I have not heard of using a temp GRUU in the From header field before.
Normally the From header field contains the AoR, not the contact URI,
and where anonymity is required if contains a value such as
sip:anonymous@example.com or a domain part of "anonymous.invalid". Of
course, a temp GRUU does provide anonymity (given that you are prepared
to disclose the domain), so I suppose it might work. There may be one or
two side effects:

1. A UAS that simply displays the From URI would display the temp GRUU
rather than something like "sip:anonymous@example.com".

2. If the outbound proxy provides an Authentication Service, it would
need to recognise that this is a temporary GRUU and sign the Identity
header field even though the From URI does not match the identity in the
Proxy-Authorization header field.

I would like to hear opinions on this.

John


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From: Aymeric Moizard <jack@atosc.org>
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To: Henry Sinnreich <hsinnrei@adobe.com>
Subject: Re: [Sip] ICE deployment data before LC for RFC
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Hi Henry,

On Mon, 16 Jul 2007, Henry Sinnreich wrote:

> In the best IETF tradition, it would also help to have open source code
> available for ICE, before declaring it a standard.

To answer one of your question: I have implemented ICE in open source
software (around version 8... not 17): It's available through oRTP and
mediastreamer2 library (not used in any open source final application)
and still miss only TURN capability.

I'm more or less waiting for an RFC before updating to the latest ICE
specification...

Regards,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/



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Subject: RE: [Sip] CFP - International Conference on IP Multimedia
	SubsystemArchitecture and Applications
Date: Tue, 17 Jul 2007 15:12:35 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Samir Saklikar" <ssaklikar@gmail.com>, <sip@ietf.org>
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On the SIP list, we generally try and discourage conference postings, =
as:

-	there are plenty of other channels for such postings to be made.
-	none of us have the bandwidth for every possible conference and =
seminar on voice over IP issues that would otherwise get posted.

We strongly suggest that if you have such a posting, you consult with =
the chairs first before making such a posting. What we want to hear =
before accepting such a posting, is how such a posting supports the =
general aims of the SIP WG. Conferences that are focussed on identifying =
and presenting new SIP techniques may well be acceptable; applying SIP =
into IMS is quite possibly not within scope.

Regards

Keith




________________________________

	From: Samir Saklikar [mailto:ssaklikar@gmail.com]=20
	Sent: Tuesday, July 17, 2007 5:32 AM
	To: sip@ietf.org; sipping@ietf.org
	Subject: [Sip] CFP - International Conference on IP Multimedia =
SubsystemArchitecture and Applications
=09
=09

	=20

	International Conference on IP Multimedia Subsystems Architecture and =
Applications-2007 (IMSAA-2007),=20
=09


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From: Robert Sparks <rjsparks@estacado.net>
Subject: Re: [Sip] Changes to the INVITE transaction to address bug 769
Date: Tue, 17 Jul 2007 09:38:40 -0500
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On Jul 17, 2007, at 1:42 AM, Nasir Khan wrote:

> Hi Robert,
>
> I have 3 comments / suggestions w.r.t this ID.
>
> Comment-1
> ----------------
> Should there be any change with respect to receiving CANCEL in  
> Accepted state? Hitherto the UAC was getting a 481/CANCEL in such  
> cases. Now we will get 200/CANCEL with no effect on IST.
> The reason for this question is backwards compatibility, however I  
> can myself argue both ways on this one. What do you think?

I don't see any issue here. The UAC already has to be prepared for  
either response.

>
>
> Comment-2
> -----------------
> In section 6.7 of the ID about the Timer-L you say - "and the  
> amount of time this (or any downstream) UAS core might be  
> retransmitting the 2xx while waiting for an ACK. "
>
> How does 2xx retransmission timer affect the choice of Timer L?
>
>
> Comment-3
> -----------------
> Shouldnt the Timer L be set to T4 which represents the amount of  
> time the network will take to clear messages between client and  
> server transactions?
> Since 2xx is sent reliably anyway, IMHO we do not need to link  
> Timer L to Timer B.

For both of these comments - Timer L is keeping the transaction in  
place while the TU on top of this server transaction finishes its
job of retrasmitting 200s until it gets an ACK. It's not just  
clearing messages that are already in the network. Remember that those
200s might get lost, and timers on the UAC's side will cause the  
INVITE request to be retransmitted. This Timer makes sure the
server transaction is still there to receive that retransmitted  
INVITE.  Does this clear up the questions you had?


>
> Thanks
>
> Nasir Khan
> BEA Systems Inc.
>
>
> On 6/27/07, Robert Sparks <rjsparks@estacado.net> wrote:
> Everyone -
>
> I've put together a proposal for addressing the spec bug captured at
> http://bugs.sipit.net/show_bug.cgi?id=769
>
> Please read through
> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
> and comment.
>
> RjS
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
> -- 
> Discuss SIP Servlets at http://groups.google.com/group/sipservlets/
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; "><BR><DIV><DIV>On Jul 17, 2007, =
at 1:42 AM, Nasir Khan wrote:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite">Hi =
Robert,<BR> <BR> I have 3 comments / suggestions w.r.t this ID. <BR> =
<BR> Comment-1<BR> ----------------<BR> Should there be any change with =
respect to receiving CANCEL in Accepted state? Hitherto the UAC was =
getting a 481/CANCEL in such cases. Now we will get 200/CANCEL with no =
effect on IST. <BR> The reason for this question is backwards =
compatibility, however I can myself argue both ways on this one. What do =
you think?<BR></BLOCKQUOTE><DIV><BR =
class=3D"khtml-block-placeholder"></DIV>I don't see any issue here. The =
UAC already has to be prepared for either =
response.=A0</DIV><DIV>=A0<BR><BLOCKQUOTE type=3D"cite"> <BR> <BR> =
Comment-2<BR> -----------------<BR> In section 6.7 of the ID about the =
Timer-L you say - "and the amount of time this (or any downstream) UAS =
core might be retransmitting the 2xx while waiting for an ACK. " <BR> =
<BR> How does 2xx retransmission timer affect the choice of Timer =
L?=A0</BLOCKQUOTE></DIV><DIV><BLOCKQUOTE type=3D"cite"> <BR> <BR> =
Comment-3<BR> -----------------<BR> Shouldnt the Timer L be set to T4 =
which represents the amount of time the network will take to clear =
messages between client and server transactions? <BR> Since 2xx is sent =
reliably anyway, IMHO we do not need to link Timer L to Timer =
B.<BR></BLOCKQUOTE><DIV><BR class=3D"khtml-block-placeholder"></DIV>For =
both of these comments - Timer L is keeping the transaction in place =
while the TU on top of this server transaction finishes =
its</DIV><DIV>job of retrasmitting 200s until it gets an ACK. It's not =
just clearing messages that are already in the network. Remember that =
those</DIV><DIV>200s might get lost, and timers on the UAC's side will =
cause the INVITE request to be retransmitted. This Timer makes sure =
the</DIV><DIV>server transaction is still there to receive that =
retransmitted INVITE.=A0 Does this clear up the questions you =
had?</DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><BR><BLOCKQUOTE =
type=3D"cite"> <BR> Thanks<BR> <BR> Nasir Khan<BR> BEA Systems Inc.<BR> =
<BR><BR><DIV><SPAN class=3D"gmail_quote">On 6/27/07, <B =
class=3D"gmail_sendername">Robert Sparks</B> &lt;<A =
href=3D"mailto:rjsparks@estacado.net">rjsparks@estacado.net</A>&gt; =
wrote:</SPAN><BLOCKQUOTE class=3D"gmail_quote" style=3D"border-left: 1px =
solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: =
1ex;"> Everyone -<BR><BR>I've put together a proposal for addressing the =
spec bug captured at<BR><A =
href=3D"http://bugs.sipit.net/show_bug.cgi?id=3D769">http://bugs.sipit.net=
/show_bug.cgi?id=3D769</A><BR><BR>Please read through<BR> <A =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt=
">http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt</A><B=
R>and =
comment.<BR><BR>RjS<BR><BR><BR>___________________________________________=
____ <BR>Sip mailing list=A0=A0<A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.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 =
developments on the application of sip<BR></BLOCKQUOTE></DIV><BR><BR =
clear=3D"all"><BR> -- <BR>Discuss SIP Servlets at <A =
href=3D"http://groups.google.com/group/sipservlets/">http://groups.google.=
com/group/sipservlets/</A><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; =
">_______________________________________________</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sip mailing list<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN><A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/=
mailman/listinfo/sip</A></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">This list is =
for NEW development of the core SIP Protocol</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Use <A =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colum=
bia.edu</A> for questions on current sip</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Use <A =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on the application of sip</DIV> =
</BLOCKQUOTE></DIV><BR></BODY></HTML>=

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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1231694194==--





From sip-bounces@ietf.org Tue Jul 17 10:43:38 2007
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Re: [Sipping] RE: Comments on
	draft-ietf-sip-multiple-refer-01.txt
Date: Tue, 17 Jul 2007 09:43:10 -0500
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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As much as I like the Referred-By mechanism, before investing too =20
much time into extending Referred-by for this case, ask yourself =20
seriously
if there is a customer for the work. AFAIK, there are no =20
implementations of the AIB part of Referred-by on the planet (please =20
speak up if you
have one) and I know of no-one who plans to build it. Unless that =20
changes, I think we'd all be better off spending our time working on =20
things
that people will actually use.

RjS


On Jul 17, 2007, at 1:34 AM, Gonzalo Camarillo wrote:

> Hi,
>
> [moving this thread from the SIPPING list to this one (SIP) because =20=

> this is a SIP document now.]
>
> You are right. It is not trivial to use Referred-By with multiple =20
> REFER. My suggestion is that, the same way as Referred-By is =20
> specified as an extension of refer, you go ahead specify (if you =20
> are interested) how to use Referred-by with multiple REFER in a =20
> different spec.
>
> The body that the referrer would add to its REFER (which is what =20
> the refer target would get) could be an XML list with the =20
> recipients (those that can be disclosed) of the multiple REFER =20
> (plus the referrer). Something similar to what recipients get after =20=

> a relay processes a list with copyControl attributes (draft-ietf-=20
> sipping-capacity-attribute-04.txt)... in any case, since multiple =20
> REFER is only used within a context of a service, you need to think =20=

> whether having a general Referred-By mechanism is needed or if such =20=

> functionality can be achieved in a different way within the =20
> services multiple REFER is used in (e.g., conferencing).
>
> Cheers,
>
> Gonzalo
>
>
> Qian Sun wrote:
>> Hello, Gonzalo
>> RFC3892 says:
>>    The sipfrag inside a Referred-By token MUST contain copies of the
>>    Refer-To, Referred-By, and Date header fields from the REFER =20
>> request.
>> In multiple refer case, the Refer-To header field includes content =20=

>> id, not the REFER-Target' URI. The content id is meaningless to =20
>> the REFER-Targets, so the Referred-By token has to include the URI-=20=

>> List.
>> Cheers,
>> Qian
>> -----Original Message-----
>> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>> Sent: Monday, July 16, 2007 4:23 PM
>> To: Qian Sun
>> Cc: Aki.Niemi@nokia.com; Markus.Isomaki@nokia.com; =20
>> miguel.an.garcia@nokia.com; hisham.khartabil@gmail.com; =20
>> sipping@ietf.org
>> Subject: Re: Comments on draft-ietf-sip-multiple-refer-01.txt
>> Hi,
>> thanks for your comments. Answers inline:
>> Qian Sun wrote:
>>  > Some comments on the draft...
>>  >
>>  > 1. I think it might be a good idea to talk something on =20
>> "Referred-By" in this draft. Referred-By mechanism is still very =20
>> useful for the multiple refer case. Maybe Refer-To related to a =20
>> URI List with "bcc" or "anonymize" is a trouble. "Referred-BY=94 in =20=

>> the message header is to identify the initial REFER-Issuer. This =20
>> information may be encrypted together with "refer-to" URI list, =20
>> including the copy control level such as to/cc/bcc to the end =20
>> user, i.e. REFER-Target, where the information will be decrypted =20
>> by the REFER-Target. In this case the REFER-Recipient won=92t be =20
>> able to get rid of the "bcc" URI from the URI list, and the "bcc" =20
>> URI will be sent to other REFER-Targets along with normal to/cc. =20
>> This will be a problem to destroy the blind cc function.
>>  > However, the REFER-issuer may opt to not include a Referred-By =20
>> token. For example:
>>  >    REFER sip:conf-123@example.com;gruu;opaque=3Dhha9s8d-999a  SIP/=20=

>> 2.0
>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>  >    Referred-By: Carol <sip:carol@chicago.example.com>
>>  >    Content-Type: application/resource-lists+xml
>>  >    Content-Disposition: recipient-list
>>  >    Content-Length: 362
>>  >    Content-ID: <cn35t8jf02@example.com>
>>  >
>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>>  >      <list>
>>  >        <entry uri=3D"sip:bill@example.com?method=3DBYE" />
>>  >        <entry uri=3D"sip:joe@example.org?method=3DBYE" />
>>  >        <entry uri=3D"sip:ted@example.net?method=3DBYE" />
>>  >      </list>
>>  >    </resource-lists>
>>  >
>>  > When the URI list has no "bcc" or "anonymize" URI, REFER-issuer =20=

>> may include the Referred-By token in the REFER body.
>> The way Referred-By works is not affected by what this draft says.
>> Therefore, I do not see the need to talk about it.
>>  >
>>  > 2. Nested refer in multiple refer is very interesting, I =20
>> believe it could be a good idea to enroll a "multiple nested =20
>> refer" example into this draft. Just for an example:
>>  >    REFER sip:list-service@example.com;gruu;opaque=3Dhha9s8d-999a  =20=

>> SIP/2.0
>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>  >    Content-ID: <cn35t8jf02@example.com>
>>  >
>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>>  >      <list>
>>  >        <entry uri=3D"sip:bill@example.com?method=3DREFER%3FRefer-To%=
=20
>> 3Dsip:conf-123%40shenzhen.example.com" />
>>  >        <entry uri=3D"sip:tom@example.com?method=3DREFER%3FRefer-To%=20=

>> 3Dsip:conf-123%40shenzhen.example.com" />
>>  >      </list>
>>  >    </resource-lists>
>>  > In this case, the REFER-Recipient acts as a Refer List service, =20=

>> has the similar function to the MESSAGE List service, i.e. =20
>> receives a REFER with URI-List and distributes some REFERs to each =20=

>> address in the URI-List.
>>  > Note that, there is only one final REFER-Target in above =20
>> "nested multiple refer" example, not multiple REFER-Targets. In =20
>> some situations, this convergent feature is useful.
>> The draft says the following:
>>     Additionally, REFER-Recipients SHOULD only accept REFER requests
>>     within the context of an application the server understands =20
>> (e.g., a
>>     conferencing application).
>> this text was added to avoid relays being used as random traffic
>> forwarders. Therefore, since a relay needs to understand what is =20
>> going
>> on (e.g., a multiple REFER to trigger BYEs towards several conference
>> attendees) and currently there is no service that uses nested =20
>> refers, I
>> do not think we need to talk about it.
>>  >
>>  > 3. I guess it might be a typo in Figure 2 "8. 200K OK", maybe =20
>> you meant "200 OK"?
>> yes, thanks for pointing it out. We will fix it.
>> Thanks,
>> Gonzalo
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Subject: [Sip] RE: [MMUSIC] ICE deployment data before LC for RFC
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> -----Original Message-----
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com] 
> Sent: Monday, July 16, 2007 3:38 PM
> To: mmusic@ietf.org; sip@ietf.org
> Cc: Cullen Jennings; jon.peterson@neustar.biz
> Subject: [MMUSIC] ICE deployment data before LC for RFC
> 
> The work on ICE is truly impressive and so are the numerous 
> I-Ds associated with ICE.
> 
>  
> 
> However, before sending the <ice-17> I-D for LC to the IESG, 
> it would be prudent and responsible to the industry (that 
> spends considerable resources on ICE in good faith) 
> implementing ICE, to either (1) make it an informational RFC 
> or 

To my knowledge, IETF only requires such deployment details and
analysis when promoting a Proposed Standard to Draft Standard,
and promoting Draft Standard to Internet Standard.

Many Internet Drafts don't have deployment results, because they
aren't yet deployed nor deployable until they achieve RFC status.
As was well witnessed by the many substantive changes to ICE during
its development, it is unreasonable to create interoperable 
implementations with an Internet Draft while that Internet Draft
is undergoing substantial and substantive revision.

I don't see the value to the community to deny ICE as a 
standards-track document.  Making ICE an Informational document,
as you suggest, will only further delay interoperation and 
deployment of ICE.  If ICE had a competing, incrementally-
deployable NAT traversal technique, I could see a desire to 
make one of those techniques a standards track document and the 
other an informational document; however, there is no such
competing, incrementally-deployable NAT traversal technique.

ICE should be a standards-track document.

> (2) publish some deployment data showing such items as:
> 
>  
> 
> *	NAT scenarios that have been tested,
> *	The % of success,
> *	Performance, such as call setup delay using SIP.
> 
>  
> 
> I have not seen any such public data on ICE deployment, for 
> example with SIP. A private report has raised my concern 
> about the performance of ICE.

I do agree that publishing information, such as what you 
requested in (2), would be very useful.  But publishing 
such information is not necessary for ICE to be published 
as a standard-track RFC.

> In the best IETF tradition, it would also help to have open 
> source code available for ICE, before declaring it a standard.

I see that already there have been two followups to your 
request for publicly-available code, pointing at three 
implementations (NICE, PJSIP, and oRTP/mediastreamer2).

> ICE is too important for the IETF (good work Jonathan and 
> other authors!) not to publish (1) deployment results and (2) 
> publish open source code as well.
> 
> The various nits discussed on the lists are only of secondary 
> importance to the LC compared to the above and would get 
> resolved with such a process anyway.

-d

> 
>  
> 
> These are my personal two cents.
> 
> Henry Sinnreich
> 
> 
> 
> 
>  
> 
> 


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Subject: [Sip] Re: [MMUSIC] ICE deployment data before LC for RFC
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Henry Sinnreich wrote:
> The work on ICE is truly impressive and so are the numerous I-Ds
> associated with ICE.
> 
> However, before sending the <ice-17> I-D for LC to the IESG, it would be
> prudent and responsible to the industry (that spends considerable
> resources on ICE in good faith) implementing ICE, to either (1) make it
> an informational RFC 

I see no reason for this work not to be standards track.

> or (2) publish some deployment data showing such
> items as:
> 
>     * NAT scenarios that have been tested,
>     * The % of success,
>     * Performance, such as call setup delay using SIP.

It is my understanding that Google Talk uses something very similar to
ICE, which at some point they will migrate to the XMPP profile of ICE
defined in XEP-0176 <http://www.xmpp.org/extensions/xep-0176.html>. They
are not typically forthcoming with detailed performance data, but the
service appears to be fairly robust and on various public discussion
lists Google Talk team members have indicated that they successfully
establish media exchange about 92+% of the time without a media relay.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/


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_______________________________________________
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From sip-bounces@ietf.org Tue Jul 17 13:56:03 2007
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Date: Tue, 17 Jul 2007 19:55:47 +0200
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Thread-Topic: [MMUSIC] ICE deployment data before LC for RFC
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, <mmusic@ietf.org>, <sip@ietf.org>
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Cc: Cullen Jennings <fluffy@cisco.com>, jon.peterson@neustar.biz
Subject: [Sip] RE: [MMUSIC] ICE deployment data before LC for RFC
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As SIP WG chair and the document shepherd for
draft-ietf-sip-ice-option-tag this is what I shall be doing with regard
to the ancillary document in this process...

I went to RFC 2026 and carefully read the section on proposed standard:


4.1.1  Proposed Standard

   The entry-level maturity for the standards track is "Proposed
   Standard".  A specific action by the IESG is required to move a
   specification onto the standards track at the "Proposed Standard"
   level.

   A Proposed Standard specification is generally stable, has resolved
   known design choices, is believed to be well-understood, has received
   significant community review, and appears to enjoy enough community
   interest to be considered valuable.  However, further experience
   might result in a change or even retraction of the specification
   before it advances.

   Usually, neither implementation nor operational experience is
   required for the designation of a specification as a Proposed
   Standard.  However, such experience is highly desirable, and will
   usually represent a strong argument in favor of a Proposed Standard
   designation.

   The IESG may require implementation and/or operational experience
   prior to granting Proposed Standard status to a specification that
   materially affects the core Internet protocols or that specifies
   behavior that may have significant operational impact on the
   Internet.

   A Proposed Standard should have no known technical omissions with
   respect to the requirements placed upon it.  However, the IESG may
   waive this requirement in order to allow a specification to advance
   to the Proposed Standard state when it is considered to be useful and
   necessary (and timely) even with known technical omissions.

   Implementors should treat Proposed Standards as immature
   specifications.  It is desirable to implement them in order to gain
   experience and to validate, test, and clarify the specification.
   However, since the content of Proposed Standards may be changed if
   problems are found or better solutions are identified, deploying
   implementations of such standards into a disruption-sensitive
   environment is not recommended.

My judgement is that at least the in terms of its draft, the SIP WG has
fulfilled the requirements for a request to publish a proposed standard.
It appears to me that any further decision about implementation rests
with the IESG, who should obviously take into account any submissions
made during the IESG last call process.

It is therefore my intent to proceed with the publish request for
draft-ietf-sip-ice-option-tag.

Regards

Keith


________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Monday, July 16, 2007 11:38 PM
	To: mmusic@ietf.org; sip@ietf.org
	Cc: Cullen Jennings; jon.peterson@neustar.biz
	Subject: [MMUSIC] ICE deployment data before LC for RFC
=09
=09

	The work on ICE is truly impressive and so are the numerous I-Ds
associated with ICE.

	=20

	However, before sending the <ice-17> I-D for LC to the IESG, it
would be prudent and responsible to the industry (that spends
considerable resources on ICE in good faith) implementing ICE, to either
(1) make it an informational RFC or (2) publish some deployment data
showing such items as:

	=20

	*	NAT scenarios that have been tested,=20
	*	The % of success,=20
	*	Performance, such as call setup delay using SIP.=20

	=20

	I have not seen any such public data on ICE deployment, for
example with SIP. A private report has raised my concern about the
performance of ICE.

	=20

	In the best IETF tradition, it would also help to have open
source code available for ICE, before declaring it a standard.

	=20

	ICE is too important for the IETF (good work Jonathan and other
authors!) not to publish (1) deployment results and (2) publish open
source code as well.

	The various nits discussed on the lists are only of secondary
importance to the LC compared to the above and would get resolved with
such a process anyway.

	=20

	These are my personal two cents.

	Henry Sinnreich
=09
=09
=09

	=20



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From sip-bounces@ietf.org Tue Jul 17 14:13:21 2007
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John;

 My comments inline..

Elwell, John wrote:
> Mayumi,
>
> "If the "temp-gruu" URI parameter and value exist within the REGISTER
>    response, the user agent SHOULD use the value of the "temp-gruu" as
>    an anonymous URI representing the originator.  This URI SHOULD be
>    used for Contact and From, for example, wherever the originator of
>    the URI is required."
> I have not heard of using a temp GRUU in the From header field before.
> Normally the From header field contains the AoR, not the contact URI,
> and where anonymity is required if contains a value such as
> sip:anonymous@example.com or a domain part of "anonymous.invalid". Of
> course, a temp GRUU does provide anonymity (given that you are prepared
> to disclose the domain), so I suppose it might work. There may be one or
> two side effects:
>
> 1. A UAS that simply displays the From URI would display the temp GRUU
> rather than something like "sip:anonymous@example.com".
>   
  I agree, this is a mistake on our side and the From URI should not 
utilize the temp-gruu, and instead
should utilize anonymous URI such as "sip:anonymous@example.com". One 
could try to use the
"anonymous.invalid" domain but this will interfere with the SIP Identity 
which heavily relies
on domain of the From header field. We will fix the text here to ensure 
temp-gruu is not
used for From URI.
> 2. If the outbound proxy provides an Authentication Service, it would
> need to recognise that this is a temporary GRUU and sign the Identity
> header field even though the From URI does not match the identity in the
> Proxy-Authorization header field.
>
>   
 If From header is populated using anonymous URI 
(anonymous@example.com),with domain portion equal to 
the domain for which the outbound proxy is responsible, this won't be 
problematic. The identity
service would use the information in the proxy-authorization to identity 
the user and will simply
judge whether that user has a right to send out request while 
withholding its identity.

 If temp-gruu is used as it is currently written, outbound proxy would 
need to look up the
location-service and see if the AoR associated with the credential is 
indeed associated with
the temp-gruu in the request. The text will be changed to populate the 
 From header field with
anonymous URI as mentioned above, so this really is not a concern anymore.

 As a side note, if there is a concern about revealing its domain name 
to the peer, one would probably
need to utilize a dedicated anonymous service(anonymity provider) which 
provides URI to be used in
 From while withholding its original domain name etc.
 
 Regards
  Shida

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



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From sip-bounces@ietf.org Tue Jul 17 14:19:02 2007
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(As WG chair)

I have just requested publication of:

http://www.ietf.org/internet-drafts/draft-ietf-sip-ice-option-tag-02.txt

The next stage for submitting comments is therefore the IESG last call.

The PROTO writeup submitted with the publication request is as follows.

Please feel free to respond if you consider any of the information
incorrect.

Regards

Keith

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

PROTO writeup for
http://www.ietf.org/internet-drafts/draft-ietf-sip-ice-option-tag-
02.txt: "Indicating Support for Interactive Connectivity Establishment
(ICE) in the=20
Session Initiation Protocol (SIP)"

   (1.a)  Who is the Document Shepherd for this document?  Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

Keith Drage

The document has been reviewed and is ready for forwarding to IESG for
publication.

   (1.b)  Has the document had adequate review both from key WG members
          and from key non-WG members?  Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

The need for a SIP option tag developed out of discussions in the MMUSIC
working group=20
on draft-ietf-sip-ice. See discussion in IETF#67, and discussions on the
IETF MMUSIC=20
list (Christer Holmber, Michael Slavitch). In conformance with RFC 3427,
this portion of=20
the work was passed to the SIP working group.

Document history:

-	draft-ietf-sip-ice-option-tag-00 was submitted 1st March 2007
and expires 1st=20
September 2007.
-	draft-ietf-sip-ice-option-tag-01 was submitted 5th March 2007
and expires 6th=20
September 2007.
-	draft-ietf-sip-ice-option-tag-02 was submitted 19th June 2007
and expires 21st=20
December 2007.

WGLC was initiated in the SIP WG on draft-ietf-sip-ice-option-tag-01 on
29th March 2007=20
with comments requested by 20th April 2007.

Review was made and comments were received from: Peter Blatherwick,
Jeroen van Bemmel,=20
Kevin Johns. During the course of the work comments have also been made
by: Ernst=20
Horvath.

   (1.c)  Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization, or XML?

The document defines mechanisms that are entirely internal to the
Session Initiation=20
Protocol (SIP). The document shepherd considers that no external review
from an external=20
specialist is necessary.

   (1.d)  Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of?  For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it.  In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here.  Has an IPR disclosure related to this document
          been filed?  If so, please include a reference to the
          disclosure and summarize the WG discussion and conclusion on
          this issue.

The document defines a new SIP protocol extension for a particular
purpose in a form=20
that has been used for many other extensions. The document shepherd has
no concerns with=20
the document.

There have been no IPR disclosures on this document.

   (1.e)  How solid is the WG consensus behind this document?  Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

The document has been reviewed by appropriate SIP experts. It is not
considered that any=20
of the topics have significant architectural or protocol impact.

   (1.f)  Has anyone threatened an appeal or otherwise indicated extreme
          discontent?  If so, please summarize the areas of conflict in
          separate email messages to the Responsible Area Director.  (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

None indicated.

   (1.g)  Has the Document Shepherd personally verified that the
          document satisfies all ID nits?  (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/.)  Boilerplate checks are
          not enough; this check needs to be thorough.  Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type, and URI type reviews?  If the document
          does not already indicate its intended status at the top of
          the first page, please indicate the intended status here.

The document has been reviewed against the guidelines in RFC 4485 and it
is believed=20
that the document is conformant with those guidelines.

For ID-NITS the checks against idnits 2.04.09 report no NITS found
except that draft-
ietf-mmusic-ice-13 is now at a later version. While this needs
correction, it should not=20
cause a problem in allowing IESG last call to proceed. There are no
dependent issues in=20
the revisions that impact the technical contents of this draft.

   (1.h)  Has the document split its references into normative and
          informative?  Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state?  If such normative references exist, what is the
          strategy for their completion?  Are there normative references
          that are downward references, as described in [RFC3967]?  If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].

The document has separate sections for normative and informative
references. The=20
normative references have been checked and found to be normative. The
informative=20
references have also been checked and found to be all informative.

There are no downrefs in the document.

   (1.i)  Has the Document Shepherd verified that the document's IANA
          Considerations section exists and is consistent with the body
          of the document?  If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries?  Are the IANA registries clearly identified?  If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations?  Does it suggest a
          reasonable name for the new registry?  See [RFC2434].  If the
          document describes an Expert Review process, has the Document
          Shepherd conferred with the Responsible Area Director so that
          the IESG can appoint the needed Expert during IESG Evaluation?

The document defines the following values that require registration:

*	A SIP option tag for ice. The IANA registry requirements are for
a standards track=20
document and this is a standards track document. The IANA considerations
section in=20
section 7.1 of the document provides for the appropriate revision to the
registry in=20
conformance with the format of that registry.
*	A media feature tag for ice. The IANA registry requirements are
for IETF consensus=20
which can here be represented by the contents of a standards track
document. The IANA=20
considerations section in section 7.2 of the document provides for the
appropriate=20
revision to the registry in conformance with the format of that
registry. Note that=20
while the presentation of the information is slightly different to that
of the IANA=20
registry, this is the format that all previous information for this
registry has been=20
presented in.

   (1.j)  Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

The document contains no material written in a formal language.

   (1.k)  The IESG approval announcement includes a Document
          Announcement Write-Up.  Please provide such a Document
          Announcement Write-Up.  Recent examples can be found in the
          "Action" announcements for approved documents.  The approval
          announcement contains the following sections:

          Technical Summary
             Relevant content can frequently be found in the abstract
             and/or introduction of the document.  If not, this may be
             an indication that there are deficiencies in the abstract
             or introduction.

          Working Group Summary
             Was there anything in the WG process that is worth noting?
             For example, was there controversy about particular points
             or were there decisions where the consensus was
             particularly rough?

          Document Quality
             Are there existing implementations of the protocol?  Have a
             significant number of vendors indicated their plan to
             implement the specification?  Are there any reviewers that
             merit special mention as having done a thorough review,
             e.g., one that resulted in important changes or a
             conclusion that the document had no substantive issues?  If
             there was a MIB Doctor, Media Type, or other Expert Review,
             what was its course (briefly)?  In the case of a Media Type
             Review, on what date was the request posted?

          Personnel
             Who is the Document Shepherd for this document?  Who is the
             Responsible Area Director?  If the document requires IANA
             experts(s), insert 'The IANA Expert(s) for the registries
             in this document are <TO BE ADDED BY THE AD>.'

Technical summary.

This specification defines a media feature tag and an option tag for use
with the=20
Session Initiation Protocol (SIP).  The media feature tag allows a UA to
communicate to=20
its registrar that it supports ICE. The option tag allows a User Agent
(UA) to require=20
support for ICE in order for a call to proceed.

Working group summary.

There is consensus in the working group to publish this document. The
document came=20
about as a result of discussions in the IETF MMUSIC working group to
progress draft-
ietf-mmusic-ice.

Document Quality

There has been no indication of implementation of this specific draft.
There are known=20
implementations of earlier versions of draft-ietf-mmusic-ice.

Personnel

The document shepherd for this document was Keith Drage. The responsible
Area Director=20
was Cullen Jennings. 'The IANA Expert(s) for the registries in this
document are <TO BE=20
ADDED BY THE AD>.

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


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From sip-bounces@ietf.org Tue Jul 17 14:50:28 2007
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Thread-Topic: Re: ICE deployment data before LC for RFC
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Subject: [Sip] Re: ICE deployment data before LC for RFC
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Following some discussions and soul searching, the best approach for
ICE-17 for LC would be to _go ahead and make it an experimental RFC_.

This would support the development of interoperable implementations, the
collection of deployment data and also hopefully open source code.

Though I have full faith in ICE, following the practices that made the
IETF successful take precedent here, I believe.

This is a change from the attached that I wrote on 7/16/2007.

Note: We can only hope the proliferation of SBC in VoIP service provider
networks will not make these concerns and ICE for SIP a moot issue, but
this is another topic.

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D

>The work on ICE is truly impressive and so are the numerous I-Ds
associated with ICE.

>However, before sending the <ice-17> I-D for LC to the IESG, it would
be prudent and responsible to the industry=20
>(that spends considerable resources on ICE in good faith) implementing
ICE, to either (1) make it an informational RFC or=20
>(2) publish some deployment data showing such items as:

>* NAT scenarios that have been tested,
>* The % of success,
>* Performance, such as call setup delay using SIP.

>I have not seen any such public data on ICE deployment, for example
with SIP.=20
>A private report has raised my concern about the performance of ICE.

>In the best IETF tradition, it would also help to have open source code
>available for ICE, before declaring it a standard.

>ICE is too important for the IETF (good work Jonathan and other
authors!) >not to publish=20
>(1) deployment results and (2) publish open source code as well.
>The various nits discussed on the lists are only of secondary
importance to >the LC=20
>compared to the above and would get resolved with such a process
anyway.

>These are my personal two cents.

Henry Sinnreich



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <mmusic@ietf.org>,
	<sip@ietf.org>
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Subject: [Sip] RE: [MMUSIC] ICE deployment data before LC for RFC
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Keith,

Thanks for the attention and sorry our mail has crossed.

Feel free to proceed as you best think. (I have made my point as well).

Henry

-----Original Message-----
From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
Sent: Tuesday, July 17, 2007 10:56 AM
To: Henry Sinnreich; mmusic@ietf.org; sip@ietf.org
Cc: Cullen Jennings; jon.peterson@neustar.biz
Subject: RE: [MMUSIC] ICE deployment data before LC for RFC

As SIP WG chair and the document shepherd for
draft-ietf-sip-ice-option-tag this is what I shall be doing with regard
to the ancillary document in this process...

I went to RFC 2026 and carefully read the section on proposed standard:


4.1.1  Proposed Standard

   The entry-level maturity for the standards track is "Proposed
   Standard".  A specific action by the IESG is required to move a
   specification onto the standards track at the "Proposed Standard"
   level.

   A Proposed Standard specification is generally stable, has resolved
   known design choices, is believed to be well-understood, has received
   significant community review, and appears to enjoy enough community
   interest to be considered valuable.  However, further experience
   might result in a change or even retraction of the specification
   before it advances.

   Usually, neither implementation nor operational experience is
   required for the designation of a specification as a Proposed
   Standard.  However, such experience is highly desirable, and will
   usually represent a strong argument in favor of a Proposed Standard
   designation.

   The IESG may require implementation and/or operational experience
   prior to granting Proposed Standard status to a specification that
   materially affects the core Internet protocols or that specifies
   behavior that may have significant operational impact on the
   Internet.

   A Proposed Standard should have no known technical omissions with
   respect to the requirements placed upon it.  However, the IESG may
   waive this requirement in order to allow a specification to advance
   to the Proposed Standard state when it is considered to be useful and
   necessary (and timely) even with known technical omissions.

   Implementors should treat Proposed Standards as immature
   specifications.  It is desirable to implement them in order to gain
   experience and to validate, test, and clarify the specification.
   However, since the content of Proposed Standards may be changed if
   problems are found or better solutions are identified, deploying
   implementations of such standards into a disruption-sensitive
   environment is not recommended.

My judgement is that at least the in terms of its draft, the SIP WG has
fulfilled the requirements for a request to publish a proposed standard.
It appears to me that any further decision about implementation rests
with the IESG, who should obviously take into account any submissions
made during the IESG last call process.

It is therefore my intent to proceed with the publish request for
draft-ietf-sip-ice-option-tag.

Regards

Keith


________________________________

	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
	Sent: Monday, July 16, 2007 11:38 PM
	To: mmusic@ietf.org; sip@ietf.org
	Cc: Cullen Jennings; jon.peterson@neustar.biz
	Subject: [MMUSIC] ICE deployment data before LC for RFC
=09
=09

	The work on ICE is truly impressive and so are the numerous I-Ds
associated with ICE.

	=20

	However, before sending the <ice-17> I-D for LC to the IESG, it
would be prudent and responsible to the industry (that spends
considerable resources on ICE in good faith) implementing ICE, to either
(1) make it an informational RFC or (2) publish some deployment data
showing such items as:

	=20

	*	NAT scenarios that have been tested,=20
	*	The % of success,=20
	*	Performance, such as call setup delay using SIP.=20

	=20

	I have not seen any such public data on ICE deployment, for
example with SIP. A private report has raised my concern about the
performance of ICE.

	=20

	In the best IETF tradition, it would also help to have open
source code available for ICE, before declaring it a standard.

	=20

	ICE is too important for the IETF (good work Jonathan and other
authors!) not to publish (1) deployment results and (2) publish open
source code as well.

	The various nits discussed on the lists are only of secondary
importance to the LC compared to the above and would get resolved with
such a process anyway.

	=20

	These are my personal two cents.

	Henry Sinnreich
=09
=09
=09

	=20



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



From sip-bounces@ietf.org Tue Jul 17 14:59:05 2007
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>, <mmusic@ietf.org>, <sip@ietf.org>
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Subject: [Sip] RE: [MMUSIC] ICE deployment data before LC for RFC
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A further word on this.

Attempting to do any status change is going to create all sorts of
downreferencing problems, which while not unresolvable, will certainly
delay getting this out.

In order to register a new SIP option tag, draft-ietf-sip-ice-option-tag
must be a standards track document. RFC 3427 mandates this.

Draft-ietf-sip-ice-option-tag normatively references
draft-ietf-mmusic-ice. If the former is standards track and the latter
is either informational or experimental, I understand that would be a
down reference.=20

I was unable to fire up the reference dependency tool to find out what
other intended standards track documents normatively referenced
draft-ietf-mmusic-ice, but I suspect there are a number.

I would therefore suggest you investigate the potential referencing
impact further if you wish to proceed.

Regards

Keith



> -----Original Message-----
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
> Sent: Tuesday, July 17, 2007 7:48 PM
> To: DRAGE, Keith (Keith); mmusic@ietf.org; sip@ietf.org
> Cc: Cullen Jennings; jon.peterson@neustar.biz; behave@ietf.org
> Subject: RE: [MMUSIC] ICE deployment data before LC for RFC
>=20
> Keith,
>=20
> Thanks for the attention and sorry our mail has crossed.
>=20
> Feel free to proceed as you best think. (I have made my point=20
> as well).
>=20
> Henry
>=20
> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> Sent: Tuesday, July 17, 2007 10:56 AM
> To: Henry Sinnreich; mmusic@ietf.org; sip@ietf.org
> Cc: Cullen Jennings; jon.peterson@neustar.biz
> Subject: RE: [MMUSIC] ICE deployment data before LC for RFC
>=20
> As SIP WG chair and the document shepherd for=20
> draft-ietf-sip-ice-option-tag this is what I shall be doing=20
> with regard to the ancillary document in this process...
>=20
> I went to RFC 2026 and carefully read the section on proposed=20
> standard:
>=20
>=20
> 4.1.1  Proposed Standard
>=20
>    The entry-level maturity for the standards track is "Proposed
>    Standard".  A specific action by the IESG is required to move a
>    specification onto the standards track at the "Proposed Standard"
>    level.
>=20
>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood,=20
> has received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>=20
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.  However, such experience is highly desirable, and will
>    usually represent a strong argument in favor of a Proposed Standard
>    designation.
>=20
>    The IESG may require implementation and/or operational experience
>    prior to granting Proposed Standard status to a specification that
>    materially affects the core Internet protocols or that specifies
>    behavior that may have significant operational impact on the
>    Internet.
>=20
>    A Proposed Standard should have no known technical omissions with
>    respect to the requirements placed upon it.  However, the IESG may
>    waive this requirement in order to allow a specification to advance
>    to the Proposed Standard state when it is considered to be=20
> useful and
>    necessary (and timely) even with known technical omissions.
>=20
>    Implementors should treat Proposed Standards as immature
>    specifications.  It is desirable to implement them in order to gain
>    experience and to validate, test, and clarify the specification.
>    However, since the content of Proposed Standards may be changed if
>    problems are found or better solutions are identified, deploying
>    implementations of such standards into a disruption-sensitive
>    environment is not recommended.
>=20
> My judgement is that at least the in terms of its draft, the=20
> SIP WG has fulfilled the requirements for a request to=20
> publish a proposed standard.
> It appears to me that any further decision about=20
> implementation rests with the IESG, who should obviously take=20
> into account any submissions made during the IESG last call process.
>=20
> It is therefore my intent to proceed with the publish request=20
> for draft-ietf-sip-ice-option-tag.
>=20
> Regards
>=20
> Keith
>=20
>=20
> ________________________________
>=20
> 	From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20
> 	Sent: Monday, July 16, 2007 11:38 PM
> 	To: mmusic@ietf.org; sip@ietf.org
> 	Cc: Cullen Jennings; jon.peterson@neustar.biz
> 	Subject: [MMUSIC] ICE deployment data before LC for RFC
> =09
> =09
>=20
> 	The work on ICE is truly impressive and so are the=20
> numerous I-Ds associated with ICE.
>=20
> 	=20
>=20
> 	However, before sending the <ice-17> I-D for LC to the=20
> IESG, it would be prudent and responsible to the industry=20
> (that spends considerable resources on ICE in good faith)=20
> implementing ICE, to either
> (1) make it an informational RFC or (2) publish some=20
> deployment data showing such items as:
>=20
> 	=20
>=20
> 	*	NAT scenarios that have been tested,=20
> 	*	The % of success,=20
> 	*	Performance, such as call setup delay using SIP.=20
>=20
> 	=20
>=20
> 	I have not seen any such public data on ICE deployment,=20
> for example with SIP. A private report has raised my concern=20
> about the performance of ICE.
>=20
> 	=20
>=20
> 	In the best IETF tradition, it would also help to have=20
> open source code available for ICE, before declaring it a standard.
>=20
> 	=20
>=20
> 	ICE is too important for the IETF (good work Jonathan and other
> authors!) not to publish (1) deployment results and (2)=20
> publish open source code as well.
>=20
> 	The various nits discussed on the lists are only of=20
> secondary importance to the LC compared to the above and=20
> would get resolved with such a process anyway.
>=20
> 	=20
>=20
> 	These are my personal two cents.
>=20
> 	Henry Sinnreich
> =09
> =09
> =09
>=20
> 	=20
>=20
>=20


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



From sip-bounces@ietf.org Tue Jul 17 15:04:03 2007
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From: "Francois Audet" <audet@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>, <sip@ietf.org>
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Thanks John.

See below (I'm removing where we are in agreement).=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: Tuesday, July 17, 2007 02:01
> To: Audet, Francois (SC100:3055); sip@ietf.org
> Subject: RE: WGLC comments on sip-sips
>=20
> Francois,
>=20
> More below. I have stripped out the comments that have been=20
> satisfactorily dealt with.
>=20
> John=20
>=20

> > > 8. In 4.1.1 "If a UA registers with a SIPS Contact header=20
> field, the=20
> > > registrar
> > >    returning a service route [RFC3608] MUST return a service route
> > >    consisting of SIP URIs if the intent of the registrar=20
> is to allow
> > >    both SIP and SIPS to be used in requests sent by that
> > client.  If a
> > >    UA registers with a SIPS Contact header field, the registrar
> > >    returning a service route MUST return a service route
> > consisting of
> > >    SIPS URIs if the intent of the registrar is to allow
> > only SIPS URIs
> > >    to be used in requests sent by that UA."
> > >
> > > Presumably the intention of this is to state that RFC 3608 places=20
> > > such a requirement on registrars. The language needs changing to=20
> > > reflect this.
> >=20
> However, you didn't respond to the second part of my comment,=20
> concerning the need for a change of language. Thinking about=20
> this further, I now see that we are in fact updating RFC 3608=20
> - correct? If so, then don't we need to state this up front,=20
> and also mention it in the Introduction section where we=20
> mention RFC 3261?

OK, I think you are right. It does in fact update 3608. I will fix it.

> >=20
> > > 11. In 4.2 "It is RECOMMENDED to use an outbound proxy as per the=20
> > > procedures
> > >    defined in [I-D.ietf-sip-outbound] for supporting UACs
> > that can not
> > >    provide a certificate for establishing a TLS connection
> > (i.e., when
> > >    server-side authentication is used)."
> > > This is a recommendation on a UA, not a proxy, so it does=20
> not belong=20
> > > as a normative statement in 4.2.
> >=20
> > I'm not sure I agree on this. Seems like the decision to use or not=20
> > the procedures of sip-outbound in a proxy is a proxy=20
> requirement, not=20
> > a UA requirement.
> [JRE] The words "It is RECOMMENDED to use an outbound proxy"=20
> sound to me like a recommendation on a UA, i.e., a=20
> recommendation that the UA be configured to use an outbound=20
> proxy. Perhaps what we are trying to say is "It is=20
> RECOMMENDED that a proxy be able to behave as an outbound=20
> proxy as per the procedures...", in which case it would=20
> indeed belong in this section.

Ok, I'll fix as per your suggestion.

> >=20
> > > 13. "When a redirect server receives a request with a SIP=20
> > > Request-URI, the
> > >    redirect server MAY redirect with a 3XX response to=20
> either a SIP=20
> > > or a
> > >    SIPS Contact header field.  If the target UAS had registered
> > >    previously using a SIPS Contact header field, the=20
> redirect server
> > >    SHOULD return a SIPS Contact header field if it is in an=20
> > > environment
> > >    where TLS is usable (as described in the previous paragraph)."
> > > This assumes that a redirect server is redirecting=20
> directly to a UA,=20
> > > i.e., to a contact. However, a redirect server can also=20
> redirect to=20
> > > another AoR that resolves to a domain proxy (or another redirect=20
> > > server). In other words, the redirect server may not have any=20
> > > knowledge of a "target UAS".
> [JRE] You didn't answer this or make any changes. Do you=20
> consider that no change is necessary?

Sorry, I missed that. I guess in that case you can redirect to whatever
you want.=20
I'll make some changes.


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Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
Date: Tue, 17 Jul 2007 16:45:58 -0500
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=20

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Thursday, July 05, 2007 15:35
> To: Audet, Francois (SC100:3055); sip@ietf.org
> Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
>=20
> Francois
>=20
> in-line
>=20
> At 11:29 AM 6/18/2007, Francois Audet wrote:
> >Hi James,
> >
> >It might be useful to have some wording on the limitations=20
> of using a=20
> >response to carry a Resource-Priority header in a response.
>=20
> Do you have any text (or direction) to offer that you would=20
> want to see in this ID?

Yes, something along the lines of the paragraph below:

> >Specifically, a UAS can not use a response to carry the=20
> >Resource-Priority in cases where the Resource-Priority "absolutely=20
> >has" to be delivered to the  UAC. It is therefore only useable for=20
> >cases where a forking proxy  may discard the response (i.e., forward=20
> >another instead) without breaking anything.
>=20
> I have to admit, I do not understand this use-case (as it is=20
> written), since I'm not sure what you are referring to when you say=20
> "..."absolutely has" to be delivered to the  UAC".  Right now, RPH=20
> isn't in any response, so when did it become a situation of=20
> "..."absolutely has" to be delivered...".

What I mean is that the error response may not be delivered to the UAC
because a forking proxy will only deliver one final response. If there
are multiple error responses on the individual forks, one being the one
with the RPH, and the other being some other response, it is up to the
proxy to decide which one to forward back.

So the UAC can not rely on receiving the RPH to "re-intiate" the session
or
provide some alternative treatment. This may or may not be a problem
depending
on the application (that's why I said "if it absolutely has").

> Remind me what HERFP is again...

HERFP is the problem described above. It stands for "Heterogeneous Error
Response Forking Problem".


> >In other words,you need to think about HERFP before using this draft.
> >
> >In some scenarios, this is probably the desired behavior.
> >
> >In other scenarios, it may not acceptable. So instead, for=20
> those scenarios,
> >the terminating side would have to send a new request with a=20
> >Resource-Priority
> >instead of using a response.

What I'm looking for is just a description of the limitation of using
a response for carrying the information, in order to alert the
implementors.

I'm not looking for a solution to the HERFP problem in this draft.

> >
> >
> > > -----Original Message-----
> > > From: James M. Polk [mailto:jmpolk@cisco.com]
> > > Sent: Saturday, June 16, 2007 14:14
> > > To: sip@ietf.org
> > > Subject: [Sip] New draft modifying RFC 4412 for Responses
> > >
> > > SIP WG
> > >
> > > Here is a new ID I've written because of implementation
> > > considerations to allow a SIP Resource-Priority header (RPH)
> > > in responses, which RFC 4412 currently disallows, assuming
> > > only that stateful devices will use RPH.  There have been
> > > many requests to have this "only stateful devices"
> > > restriction relaxed, in certain scenarios.  This ID does
> > > this, and proposes how IANA is changed accordingly wrt RPH.
> > > This ID does nothing else.
> > >
> > > Comments are appreciated.
> > >
> > > >A New Internet-Draft is available from the on-line=20
> Internet-Drafts
> > > >directories.
> > > >
> > > >         Title           : Allowing SIP Resource=20
> Priority Header in
> > > > SIP Responses
> > > >         Author(s)       : J. Polk
> > > >         Filename        : draft-polk-sip-rph-in-responses-00.txt
> > > >         Pages           : 6
> > > >         Date            : 2007-6-15
> > > >
> > > >    The Session Initiation Protocol (SIP) Resource=20
> Priority Header
> > > >    (RPH), in its current form, is ignored in SIP responses.
> > >  This was a
> > > >    design choice during RFC 4412's development. This is now
> > > considered
> > > >    a bad design choice in certain scenarios.  This=20
> document corrects
> > > >    RFC 4412's communications model by optionally allowing a
> > > SIP server
> > > >    or user agent client to process the=20
> Resource-Priority Header in a
> > > >    response.
> > > >
> > > >A URL for this Internet-Draft is:
> > > >http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-res
> > > ponses-00.
> > > >txt
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the=20
> application of sip
> > >
>=20


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"Francois Audet" <audet@nortel.com> wrote on 07/17/2007 05:45:58 PM:

> 
> What I mean is that the error response may not be delivered to the UAC
> because a forking proxy will only deliver one final response. If there
> are multiple error responses on the individual forks, one being the one
> with the RPH, and the other being some other response, it is up to the
> proxy to decide which one to forward back.
> 
> So the UAC can not rely on receiving the RPH to "re-intiate" the session
> or
> provide some alternative treatment. This may or may not be a problem
> depending
> on the application (that's why I said "if it absolutely has").
> 

I don't think that will be a problem.

For one thing (for the namespaces I am working with) the expectation would 
be that ALL the responses (except 100 and 403) would have the RPH on the 
response, so it wouldn't matter which one made it back to the UAC.

Secondly, the expectation is that the devices responsible for 
"re-initiating" WOULD be stateful, and not dependant on the information in 
the RPH in the response.  It is the intervening devices that might not be 
stateful, and would take advantage of the RPH information in the 
responses.

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

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<br><font size=2 face="sans-serif"><br>
<br>
</font>
<br><font size=2><tt>&quot;Francois Audet&quot; &lt;audet@nortel.com&gt;
wrote on 07/17/2007 05:45:58 PM:<br>
<br>
&gt; <br>
&gt; What I mean is that the error response may not be delivered to the
UAC<br>
&gt; because a forking proxy will only deliver one final response. If there<br>
&gt; are multiple error responses on the individual forks, one being the
one<br>
&gt; with the RPH, and the other being some other response, it is up to
the<br>
&gt; proxy to decide which one to forward back.<br>
&gt; <br>
&gt; So the UAC can not rely on receiving the RPH to &quot;re-intiate&quot;
the session<br>
&gt; or<br>
&gt; provide some alternative treatment. This may or may not be a problem<br>
&gt; depending<br>
&gt; on the application (that's why I said &quot;if it absolutely has&quot;).<br>
&gt; </tt></font>
<br>
<br><font size=2><tt>I don't think that will be a problem.</tt></font>
<br>
<br><font size=2><tt>For one thing (for the namespaces I am working with)
the expectation would be that ALL the responses (except 100 and 403) would
have the RPH on the response, so it wouldn't matter which one made it back
to the UAC.</tt></font>
<br>
<br><font size=2><tt>Secondly, the expectation is that the devices responsible
for &quot;re-initiating&quot; WOULD be stateful, and not dependant on the
information in the RPH in the response. &nbsp;It is the intervening devices
that might not be stateful, and would take advantage of the RPH information
in the responses.</tt></font>
<br>
<br><font size=2><tt>Janet<br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
</tt></font>
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_______________________________________________
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Date: Wed, 18 Jul 2007 11:29:29 +0800
From: Qian Sun <sunqian@huawei.com>
Subject: RE: [Sip] Comments on draft-ietf-sip-subnot-etags-00
In-reply-to: <469C9168.4060907@nokia.com>
To: 'Aki Niemi' <aki.niemi@nokia.com>
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Hi Aki,

As I know, OMA PAG has adopted both filtering and this subnot-etags, maybe some companies are developing their products according to OMA specification. 

Cheers,
Qian

-----Original Message-----
From: Aki Niemi [mailto:aki.niemi@nokia.com] 
Sent: Tuesday, July 17, 2007 5:53 PM
To: ext Qian Sun
Cc: 'IETF SIP List'
Subject: Re: [Sip] Comments on draft-ietf-sip-subnot-etags-00

Hi Qian,

And sorry for a late answer to your comment. I have been on vacation
enjoying the beautiful Finnish summer. ;)

ext Qian Sun wrote:
> Hi, I am not sure whether the following description in this draft is
> enough for the usage of Entity-tags with filtering.
> 
> "5.1.  Generating Entity-tags The views might correspond to different
> groups of users that have varying levels of access rights to the
> resource state, or to subscribers that have modified their
> subscription using event notification filtering [11]."
> 
> My consideration is: If a subscriber has a subscription with filter
> F1, the current corresponding entity-tag is E1. Then the same
> subscriber MUST NOT send SUBSCRIBE with a new filter F2 and the
> entity-tag E1 in the conditional header field. i.e., if a subscriber
> changes filter for an event state, the Suppress-XXX-If-Match should
> not be used. The client of subscriber has to record the last filter
> for event state and compare filters.

If I understand your concern correctly, the problem is with filtering
and not with subnot-etags. I think a subscription that installs a
filter, or disables/removes it could effectively be an action that
itself changes the event state, and causes an if-match condition to fail.

But this all depends on the actual implementation of the filtering
system. In other words, this seems like an implementation detail of the
filtering system.

At the moment, I am not aware of any existing filtering implementations,
so I'm reluctant to adding any specific text to the draft either. If
this becomes a problem, we can always write an update to the filtering
specifications.

Cheers,
Aki




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From sip-bounces@ietf.org Wed Jul 18 02:54:31 2007
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From: Aki Niemi <aki.niemi@nokia.com>
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To: ext Qian Sun <sunqian@huawei.com>
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Hi Qian,

ext Qian Sun wrote:
> Hi Aki,
> 
> As I know, OMA PAG has adopted both filtering and this subnot-etags,
> maybe some companies are developing their products according to OMA
> specification.

If that's the case then would this be something for which the OMA PAG
specifications could maybe provide some clarification?

Cheers,
Aki


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From sip-bounces@ietf.org Wed Jul 18 04:10:57 2007
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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Henry Sinnreich skrev:
> Following some discussions and soul searching, the best approach for
> ICE-17 for LC would be to _go ahead and make it an experimental RFC_.
> 
> This would support the development of interoperable implementations, the
> collection of deployment data and also hopefully open source code.
> 
> Though I have full faith in ICE, following the practices that made the
> IETF successful take precedent here, I believe.
> 
> This is a change from the attached that I wrote on 7/16/2007.
> 
> Note: We can only hope the proliferation of SBC in VoIP service provider
> networks will not make these concerns and ICE for SIP a moot issue, but
> this is another topic.

Henry,

I don't quite understand why you are requesting a status change of ICE? 
To me it appears that ICE is almost finished spec. It has been 
implemented, used and feedback has been influencing the protocol. It is 
definitly one of the better examples of current IETF work that actually 
has running code. Sure, the implementations may not be fully updated yet 
to the latest draft version. But I think it is safe to say that ICE 
works in the intended core cases. If there are some corner cases where 
it fails, that may be. But that is not information we will have until 
really large scale deployement or someone makes very extensive tests.

I think the IETF and the Internet has much more benefit from a standards 
track solution than any of the risks existing with ICE of today.

And to be clear, proposed standard is allowed to be published without a 
single implementation exists. It might not be good practice, but it is 
allowed by our process. And as I said before ICE has had enough 
implementation that I am very confortable in balloting YES for it when 
it appears on the IESGs table.

Cheers

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM/M
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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From sip-bounces@ietf.org Wed Jul 18 07:30:46 2007
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sip] Draft slides for body (MIME) handling
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Folks,

here you have the slides I have put together for the body (MIME) 
handling presentation on Monday:

http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt

These slides relate to this draft:

http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt

The slides discuss all the open issues that have been brought up in the 
list in the last months.

Comments are welcome.

Thanks,

Gonzalo



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From sip-bounces@ietf.org Wed Jul 18 10:20:56 2007
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
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Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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Magnus,

=20

I sincerely appreciate your interest and this note!

=20

> I don't quite understand why you are requesting a status change of ICE

=20

However the arguments stated by the WG co-chairs and by you here are
procedural in nature, whereas the facts are some cause for serious
concern, and I will just mention a few facts:

=20

RFC 3489 (STUN) was also presented as a NAT traversal solution back in
2003, but <draft-ietf-behave-rfc3489bis-07> now clearly admits that
"STUN is not a NAT traversal solution by itself. Rather, it is a tool to
be used in the context of a NAT traversal solution. This is an important
change from the previous version of this specification (RFC 3489), which
presented STUN as a complete solution."

http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-07.txt

My guess is, in plain English, RFC 3489 turned out just NOT to be an
acceptable solution as it was hoped to be.=20

=20

The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the ICE
call flow with two endpoints behind two NAT (Fig. 22 on page 38) require
66 messages just for the SIP INVITE!

http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-07.
txt=20

What happens if the two endpoints are in different ISPs that have NAT of
their own in addition to the ones in Fig. 22?

What happens if one would try ICE along a P2PSIP route with say 5-8
hops? Would there be hundreds of messages? What is the likelihood for
failure?

=20

By contrast with such approaches based on hope and faith, the BEHAVE WG
has produced work based on real measurements and tools to do the
measurements, such as

=20

"NAT Classification Test Results"

=20

"Application Design Guidelines for Traversal through Network Address
Translators"=20

The measurements for hole punching are reported in

"Peer-to-Peer Communication Across Network Address Translators"=20

http://www.brynosaurus.com/pub/net/p2pnat/

=20

I am also a believer in ICE and have actually urged Jonathan Rosenberg
to do this work and he has given his best. Now that we have ICE-17
version, it is in the best interest of the industry and the IETF to make
it an experimental RFC, but not yet a standard so that:

 1. Independent, compliant implementations will be developed,

 2. Testing in events such as SIPit,

 3. Reporting the results, such as reported for the "hole punching"
above.

=20

What happens if indeed hole punching proves to be the far more effective
approach?

=20

The industry can ill afford a new series of RFC[ICE]-bis and all the
years it takes to move from believing to measuring and proving. I am
sure IETF procedures were meant in this spirit (though I am not good in
producing the right quotes).

=20

Thanks again for your attention to this concern about ICE.=20

Also my apology to all the anguish to the numerous contributors to the
ICE related I-Ds, but writing test tools and reporting measurements is
the right way to go.

=20

Henry

=20

=20

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Wednesday, July 18, 2007 1:11 AM
To: Henry Sinnreich
Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC

=20

Henry Sinnreich skrev:

> Following some discussions and soul searching, the best approach for

> ICE-17 for LC would be to _go ahead and make it an experimental RFC_.

>=20

> This would support the development of interoperable implementations,
the

> collection of deployment data and also hopefully open source code.

>=20

> Though I have full faith in ICE, following the practices that made the

> IETF successful take precedent here, I believe.

>=20

> This is a change from the attached that I wrote on 7/16/2007.

>=20

> Note: We can only hope the proliferation of SBC in VoIP service
provider

> networks will not make these concerns and ICE for SIP a moot issue,
but

> this is another topic.

=20

Henry,

=20

I don't quite understand why you are requesting a status change of ICE?=20

To me it appears that ICE is almost finished spec. It has been=20

implemented, used and feedback has been influencing the protocol. It is=20

definitly one of the better examples of current IETF work that actually=20

has running code. Sure, the implementations may not be fully updated yet


to the latest draft version. But I think it is safe to say that ICE=20

works in the intended core cases. If there are some corner cases where=20

it fails, that may be. But that is not information we will have until=20

really large scale deployement or someone makes very extensive tests.

=20

I think the IETF and the Internet has much more benefit from a standards


track solution than any of the risks existing with ICE of today.

=20

And to be clear, proposed standard is allowed to be published without a=20

single implementation exists. It might not be good practice, but it is=20

allowed by our process. And as I said before ICE has had enough=20

implementation that I am very confortable in balloting YES for it when=20

it appears on the IESGs table.

=20

Cheers

=20

Magnus Westerlund

=20

IETF Transport Area Director & TSVWG Chair

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

Multimedia Technologies, Ericsson Research EAB/TVM/M

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

Ericsson AB                | Phone +46 8 4048287

Torshamsgatan 23           | Fax   +46 8 7575550

S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com

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


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<div class=3DSection1>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Magnus,<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I sincerely appreciate your interest and this =
note!<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; I don't quite understand why you are requesting a status =
change of
ICE<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>However the a<st1:PersonName =
w:st=3D"on">rg</st1:PersonName>uments stated
by the WG co-chairs and by you here are procedural in nature, whereas =
the facts
are some cause for serious concern, and I will just mention a few =
facts:<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>RFC 3489 (STUN) was also presented as a NAT traversal solution =
back in
2003, but &lt;draft-ietf-behave-rfc3489bis-07&gt; now clearly admits =
that &quot;STUN
is not a NAT traversal solution by itself. Rather, it is a tool to be =
used in
the context of a NAT traversal solution. This is an important change =
from the
previous version of this specification (RFC 3489), which presented STUN =
as a
complete solution.&quot;<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-=
07.txt">http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-=
07.txt</a><o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>My guess is, in plain English, RFC 3489 turned out just NOT to =
be an
acceptable solution as it was hoped to be. <o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The excellent I-D &lt;draft-ietf-sipping-nat-scenarios-07&gt; =
shows the
ICE call flow with two endpoints behind two NAT (Fig. 22 on page 38) =
require 66
messages just for the SIP INVITE!<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenar=
ios-07.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-sc=
enarios-07.txt</a>
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if the two endpoints are in different ISPs that =
have NAT
of their own in addition to the ones in Fig. =
22?<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if one would try ICE along a P2PSIP route with say =
5-8
hops? Would there be hundreds of messages? What is the likelihood for =
failure?<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>By contrast with such approaches based on hope and faith, the =
BEHAVE WG
has produced work based on real measurements and tools to do the =
measurements,
such as<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;NAT Classification Test =
Results&quot;<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;Application Design Guidelines for Traversal through =
Network
Address Translators&quot; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The measurements for hole punching are reported =
in<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;Peer-to-Peer Communication Across Network Address
Translators&quot; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a =
href=3D"http://www.brynosaurus.com/pub/net/p2pnat/">http://www.brynosauru=
s.com/pub/net/p2pnat/</a><o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I am also a believer in ICE and have actually u<st1:PersonName =
w:st=3D"on">rg</st1:PersonName>ed
Jonathan Rosenbe<st1:PersonName w:st=3D"on">rg</st1:PersonName> to do =
this work
and he has given his best. Now that we have ICE-17 version, it is in the =
best interest
of the industry and the IETF to make it an experimental RFC, but not yet =
a
standard so that:<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;1. Independent, compliant implementations will be =
developed,<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;2. Testing in events such as =
SIPit,<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;3. Reporting the results, such as reported for the =
&quot;hole
punching&quot; above.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if indeed hole punching proves to be the far more
effective approach?<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The industry can ill afford a new series of RFC[ICE]-bis and all =
the
years it takes to move from believing to measuring and proving. I am =
sure IETF
procedures were meant in this spirit (though I am not good in producing =
the
right quotes).<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Thanks again for your attention to this concern about ICE. =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Also my apology to all the anguish to the numerous contributors =
to the
ICE related I-Ds, but writing test tools and reporting measurements is =
the
right way to go.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Henry<o:p></o:p></span></font></p>

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

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----Original Message-----<br>
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] <br>
Sent: Wednesday, July 18, 2007 1:11 AM<br>
To: <st1:PersonName w:st=3D"on">Henry Sinnreich</st1:PersonName><br>
Cc: <st1:PersonName w:st=3D"on">mmusic@ietf.o<st1:PersonName =
w:st=3D"on">rg</st1:PersonName></st1:PersonName>;
sip@ietf.o<st1:PersonName w:st=3D"on">rg</st1:PersonName>; =
behave@ietf.o<st1:PersonName
w:st=3D"on">rg</st1:PersonName><br>
Subject: Re: [BEHAVE] Re: ICE deployment data before LC for =
RFC</span></font></p>

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

<p class=3DMsoPlainText><st1:PersonName w:st=3D"on"><font size=3D2 =
face=3D"Courier New"><span
 style=3D'font-size:10.0pt'>Henry =
Sinnreich</span></font></st1:PersonName> skrev:<o:p></o:p></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Following some discussions and soul searching, the best =
approach
for<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; ICE-17 for LC would be to _go ahead and make it an =
experimental
RFC_.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; This would support the development of interoperable
implementations, the<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; collection of deployment data and also hopefully open =
source code.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Though I have full faith in ICE, following the practices =
that made
the<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; IETF successful take precedent here, I =
believe.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; This is a change from the attached that I wrote on =
7/16/2007.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Note: We can only hope the proliferation of SBC in VoIP =
service
provider<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; networks will not make these concerns and ICE for SIP a =
moot
issue, but<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; this is another topic.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Henry,<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I don't quite understand why you are requesting a status change =
of ICE?
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>To me it appears that ICE is almost finished spec. It has been =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>implemented, used and feedback has been influencing the =
protocol. It is
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>definitly one of the better examples of current IETF work that =
actually
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>has running code. Sure, the implementations may not be fully =
updated
yet <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>to the latest draft version. But I think it is safe to say that =
ICE <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>works in the intended core cases. If there are some corner cases =
where <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>it fails, that may be. But that is not information we will have =
until <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>really la<st1:PersonName w:st=3D"on">rg</st1:PersonName>e scale
deployement or someone makes very extensive =
tests.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I think the IETF and the Internet has much more benefit from a
standards <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>track solution than any of the risks existing with ICE of =
today.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>And to be clear, proposed standard is allowed to be published =
without a
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>single implementation exists. It might not be good practice, but =
it is <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>allowed by our process. And as I said before ICE has had enough =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>implementation that I am very confortable in balloting YES for =
it when <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>it appears on the IESGs table.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Cheers<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Magnus Westerlund<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>IETF Transport Area Director &amp; TSVWG =
Chair<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----------------------------------------------------------------=
-----<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Multimedia Technologies, Ericsson Research =
EAB/TVM/M<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----------------------------------------------------------------=
-----<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><st1:place w:st=3D"on"><st1:City =
w:st=3D"on"><font size=3D2
  face=3D"Courier New"><span =
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_______________________________________________
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Thanks for sending out in advance. Maybe we can get into the discussion 
of the slides now, before the meeting.

Regarding references to body parts:

IMO it is a *bad* idea to make the decision about processing of a body 
part "on its own" dependent on whether the node doing so has detected a 
reference to that body part or not.

For one thing, that requires a multi-pass parsing mechanism to check for 
references before deciding what processing to do on a particular part. 
For another, it requires that the recipient understand every reference. 
For instance, the sender may have included an extension header that 
contains a reference to a body part. If the extension isn't understood, 
then the body part should not be processed. But if the recipient doesn't 
understand the extension then it won't recognize the reference either, 
and so will attempt to process the body part as one without references.

The only good solution I see to this is to partition the 
Content-Disposition and/or Content-Type values whose processing is 
dependent on reference from those whose processing is not dependent on a 
reference. For instance, we could simply define a new 
Content-Disposition of "by-reference".

Then a normal sequence of processing would be to first process all the 
body parts according to their type and disposition alone. Any with 
disposition "by-reference" would be set aside. Then, when processing 
headers, if a header with a reference is encountered the appropriate 
processing could be applied to the set aside body part. References 
within other body parts would be a little more complex, but you get the 
idea.

This would not prevent reference to body parts with other dispositions. 
It just means that those body parts would also be processed according to 
their type and disposition, in addition to any processing they might get 
according to their reference. The "by-reference" disposition is just a 
defintion that gets no implicit processing.

An advantage of this approach is that by-reference body parts don't need 
to be parsed unless the reference is processed and the action required 
in processing the reference requires the node to parse the referenced 
body. (Interaction with handling=required in this case is TBD.) Even a 
multipart with disposition of by-reference might not need to be parsed.

Regarding Content-Disposition in multiparts:

While it seems "wrong" to me, I can live with using "render" as the 
normal disposition for multipart, *if* we can do a careful job of 
defining what "render" actually means for SIP. Used this way it clearly 
doesn't always mean to display the content to a user.

	Thanks,
	Paul

Gonzalo Camarillo wrote:
> Folks,
> 
> here you have the slides I have put together for the body (MIME) 
> handling presentation on Monday:
> 
> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt
> 
> These slides relate to this draft:
> 
> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
> 
> 
> The slides discuss all the open issues that have been brought up in the 
> list in the last months.
> 
> Comments are welcome.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jul 18 10:53:47 2007
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To: "'Henry Sinnreich'" <hsinnrei@adobe.com>,
	"'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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Greetings,

 

What I think Henry is advocating is a systematic approach to making sure
that the content of the ICE spec is right for its intended purpose, i.e. a
complete, general NAT traversal solution.  NAT traversal has been in my
opinion one of the most difficult aspects of SIP implementation and its very
important that its eventual standardized solution be right.  I know I spent
a lot of time implementing all the "tests" in the first 3489, only to
discover that they were not that useful.  There are two valid technical
concerns that are put forth in this email:

 

1)     Viability of correct operation in practical implementations.  ICE can
result in very complicated message transfers.

2)     Performance.  As pointed out in this email and in other places, the
additional latency associated with ICE operations can be significant.

 

Henry also points out that there are competing approaches (e.g. Hole
Punching) that can perform well in many situations and have better
performance.  IMHO, another approach by the IETF in general would be to
standardize on Hole Punching and leverage the IETF's considerable influence
to force NAT vendors to migrate towards implementations that are friendly to
Hole Punching.

 

The bottom line is that I agree with Henry in that labeling ICE as an
experimental draft is the right thing to do.  It will allow those
implementing ICE (including myself in all likelihood) to proceed but signals
to the users the correct status of the standard's contents.

 

I also want to applaud loudly the underlying tone of Henry's thread here.
That is, in solving specific technical problems within the IETF forum,
focusing on empirical observation of working implementations should be the
MO that we all strive for when developing and proposing what may become
protocol standards.

 

Thanks,

FM

 

 

  _____  

From: Henry Sinnreich [mailto:hsinnrei@adobe.com] 
Sent: Wednesday, July 18, 2007 8:20 AM
To: Magnus Westerlund
Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC

 

Magnus,

 

I sincerely appreciate your interest and this note!

 

> I don't quite understand why you are requesting a status change of ICE

 

However the arguments stated by the WG co-chairs and by you here are
procedural in nature, whereas the facts are some cause for serious concern,
and I will just mention a few facts:

 

RFC 3489 (STUN) was also presented as a NAT traversal solution back in 2003,
but <draft-ietf-behave-rfc3489bis-07> now clearly admits that "STUN is not a
NAT traversal solution by itself. Rather, it is a tool to be used in the
context of a NAT traversal solution. This is an important change from the
previous version of this specification (RFC 3489), which presented STUN as a
complete solution."

http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-07.txt

My guess is, in plain English, RFC 3489 turned out just NOT to be an
acceptable solution as it was hoped to be. 

 

The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the ICE call
flow with two endpoints behind two NAT (Fig. 22 on page 38) require 66
messages just for the SIP INVITE!

http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-07.txt 

What happens if the two endpoints are in different ISPs that have NAT of
their own in addition to the ones in Fig. 22?

What happens if one would try ICE along a P2PSIP route with say 5-8 hops?
Would there be hundreds of messages? What is the likelihood for failure?

 

By contrast with such approaches based on hope and faith, the BEHAVE WG has
produced work based on real measurements and tools to do the measurements,
such as

 

"NAT Classification Test Results"

 

"Application Design Guidelines for Traversal through Network Address
Translators" 

The measurements for hole punching are reported in

"Peer-to-Peer Communication Across Network Address Translators" 

http://www.brynosaurus.com/pub/net/p2pnat/

 

I am also a believer in ICE and have actually urged Jonathan Rosenberg to do
this work and he has given his best. Now that we have ICE-17 version, it is
in the best interest of the industry and the IETF to make it an experimental
RFC, but not yet a standard so that:

 1. Independent, compliant implementations will be developed,

 2. Testing in events such as SIPit,

 3. Reporting the results, such as reported for the "hole punching" above.

 

What happens if indeed hole punching proves to be the far more effective
approach?

 

The industry can ill afford a new series of RFC[ICE]-bis and all the years
it takes to move from believing to measuring and proving. I am sure IETF
procedures were meant in this spirit (though I am not good in producing the
right quotes).

 

Thanks again for your attention to this concern about ICE. 

Also my apology to all the anguish to the numerous contributors to the ICE
related I-Ds, but writing test tools and reporting measurements is the right
way to go.

 

Henry

 

 

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
Sent: Wednesday, July 18, 2007 1:11 AM
To: Henry Sinnreich
Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC

 

Henry Sinnreich skrev:

> Following some discussions and soul searching, the best approach for

> ICE-17 for LC would be to _go ahead and make it an experimental RFC_.

> 

> This would support the development of interoperable implementations, the

> collection of deployment data and also hopefully open source code.

> 

> Though I have full faith in ICE, following the practices that made the

> IETF successful take precedent here, I believe.

> 

> This is a change from the attached that I wrote on 7/16/2007.

> 

> Note: We can only hope the proliferation of SBC in VoIP service provider

> networks will not make these concerns and ICE for SIP a moot issue, but

> this is another topic.

 

Henry,

 

I don't quite understand why you are requesting a status change of ICE? 

To me it appears that ICE is almost finished spec. It has been 

implemented, used and feedback has been influencing the protocol. It is 

definitly one of the better examples of current IETF work that actually 

has running code. Sure, the implementations may not be fully updated yet 

to the latest draft version. But I think it is safe to say that ICE 

works in the intended core cases. If there are some corner cases where 

it fails, that may be. But that is not information we will have until 

really large scale deployement or someone makes very extensive tests.

 

I think the IETF and the Internet has much more benefit from a standards 

track solution than any of the risks existing with ICE of today.

 

And to be clear, proposed standard is allowed to be published without a 

single implementation exists. It might not be good practice, but it is 

allowed by our process. And as I said before ICE has had enough 

implementation that I am very confortable in balloting YES for it when 

it appears on the IESGs table.

 

Cheers

 

Magnus Westerlund

 

IETF Transport Area Director & TSVWG Chair

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

Multimedia Technologies, Ericsson Research EAB/TVM/M

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

Ericsson AB                | Phone +46 8 4048287

Torshamsgatan 23           | Fax   +46 8 7575550

S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com

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


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<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>Greetings,<o:p></o:p></span></font><=
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<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
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<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>What I think Henry is advocating is =
a
systematic approach to making sure that the content of the ICE spec is =
right
for its intended purpose, i.e. a complete, general NAT traversal =
solution.&nbsp;
NAT traversal has been in my opinion one of the most difficult aspects =
of SIP
implementation and its very important that its eventual standardized =
solution
be right.&nbsp; I know I spent a lot of time implementing all the =
&#8220;tests&#8221;
in the first 3489, only to discover that they were not that =
useful.&nbsp; There
are two valid technical concerns that are put forth in this =
email:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:.5in;text-indent:-.25in;mso-list:l0 level1 =
lfo1'><![if !supportLists]><font
size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:12.0pt;font-family:Arial;
color:blue'><span style=3D'mso-list:Ignore'>1)<font size=3D1 =
face=3D"Times New Roman"><span
style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp; =
</span></font></span></span></font><![endif]><font
color=3Dblue face=3DArial><span =
style=3D'font-family:Arial;color:blue'>Viability of correct
operation in practical implementations.&nbsp; ICE can result in very
complicated message transfers.<o:p></o:p></span></font></p>

<p class=3DMsoNormal =
style=3D'margin-left:.5in;text-indent:-.25in;mso-list:l0 level1 =
lfo1'><![if !supportLists]><font
size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:12.0pt;font-family:Arial;
color:blue'><span style=3D'mso-list:Ignore'>2)<font size=3D1 =
face=3D"Times New Roman"><span
style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp; =
</span></font></span></span></font><![endif]><font
color=3Dblue face=3DArial><span =
style=3D'font-family:Arial;color:blue'>Performance.&nbsp;
As pointed out in this email and in other places, the additional latency
associated with ICE operations can be =
significant.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>Henry also points out that there =
are
competing approaches (e.g. Hole Punching) that can perform well in many
situations and have better performance.&nbsp; IMHO, another approach by =
the
IETF in general would be to standardize on Hole Punching and leverage =
the IETF&#8217;s
considerable influence to force NAT vendors to migrate towards =
implementations
that are friendly to Hole Punching.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>The bottom line is that I agree =
with Henry
in that labeling ICE as an experimental draft is the right thing to =
do.&nbsp;
It will allow those implementing ICE (including myself in all =
likelihood) to proceed
but signals to the users the correct status of the standard&#8217;s =
contents.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>I also want to applaud loudly the
underlying tone of Henry&#8217;s thread here.&nbsp; That is, in solving
specific technical problems within the IETF forum, focusing on empirical =
observation
of working implementations should be the MO that we all strive for when
developing and proposing what may become protocol =
standards.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>Thanks,<o:p></o:p></span></font></p>=


<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'>FM<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3DArial><span =
style=3D'font-size:
12.0pt;font-family:Arial;color:blue'><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'> Henry =
Sinnreich
[mailto:hsinnrei@adobe.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, July 18, =
2007
8:20 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Magnus Westerlund<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org; =
behave@ietf.org;
mmusic@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] RE: =
[BEHAVE] Re:
ICE deployment data before LC for RFC</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=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Magnus,<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I sincerely appreciate your interest and this =
note!<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; I don't quite understand why you are requesting a status =
change of
ICE<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>However the a<st1:PersonName =
w:st=3D"on">rg</st1:PersonName>uments stated
by the WG co-chairs and by you here are procedural in nature, whereas =
the facts
are some cause for serious concern, and I will just mention a few =
facts:<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>RFC 3489 (STUN) was also presented as a NAT traversal solution =
back in
2003, but &lt;draft-ietf-behave-rfc3489bis-07&gt; now clearly admits =
that
&quot;STUN is not a NAT traversal solution by itself. Rather, it is a =
tool to
be used in the context of a NAT traversal solution. This is an important =
change
from the previous version of this specification (RFC 3489), which =
presented
STUN as a complete solution.&quot;<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-=
07.txt">http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-=
07.txt</a><o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>My guess is, in plain English, RFC 3489 turned out just NOT to =
be an
acceptable solution as it was hoped to be. <o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The excellent I-D &lt;draft-ietf-sipping-nat-scenarios-07&gt; =
shows the
ICE call flow with two endpoints behind two NAT (Fig. 22 on page 38) =
require 66
messages just for the SIP INVITE!<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenar=
ios-07.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-sc=
enarios-07.txt</a>
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if the two endpoints are in different ISPs that =
have NAT
of their own in addition to the ones in Fig. =
22?<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if one would try ICE along a P2PSIP route with say =
5-8
hops? Would there be hundreds of messages? What is the likelihood for =
failure?<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>By contrast with such approaches based on hope and faith, the =
BEHAVE WG
has produced work based on real measurements and tools to do the =
measurements,
such as<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;NAT Classification Test =
Results&quot;<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;Application Design Guidelines for Traversal through =
Network
Address Translators&quot; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The measurements for hole punching are reported =
in<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&quot;Peer-to-Peer Communication Across Network Address
Translators&quot; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'><a =
href=3D"http://www.brynosaurus.com/pub/net/p2pnat/">http://www.brynosauru=
s.com/pub/net/p2pnat/</a><o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I am also a believer in ICE and have actually u<st1:PersonName =
w:st=3D"on">rg</st1:PersonName>ed
Jonathan Rosenbe<st1:PersonName w:st=3D"on">rg</st1:PersonName> to do =
this work
and he has given his best. Now that we have ICE-17 version, it is in the =
best
interest of the industry and the IETF to make it an experimental RFC, =
but not
yet a standard so that:<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;1. Independent, compliant implementations will be =
developed,<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;2. Testing in events such as =
SIPit,<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&nbsp;3. Reporting the results, such as reported for the =
&quot;hole
punching&quot; above.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>What happens if indeed hole punching proves to be the far more
effective approach?<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>The industry can ill afford a new series of RFC[ICE]-bis and all =
the
years it takes to move from believing to measuring and proving. I am =
sure IETF
procedures were meant in this spirit (though I am not good in producing =
the
right quotes).<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Thanks again for your attention to this concern about ICE. =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Also my apology to all the anguish to the numerous contributors =
to the
ICE related I-Ds, but writing test tools and reporting measurements is =
the
right way to go.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Henry<o:p></o:p></span></font></p>

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

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----Original Message-----<br>
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] <br>
Sent: Wednesday, July 18, 2007 1:11 AM<br>
To: <st1:PersonName w:st=3D"on">Henry Sinnreich</st1:PersonName><br>
Cc: <st1:PersonName w:st=3D"on">mmusic@ietf.o<st1:PersonName =
w:st=3D"on">rg</st1:PersonName></st1:PersonName>;
sip@ietf.o<st1:PersonName w:st=3D"on">rg</st1:PersonName>; =
behave@ietf.o<st1:PersonName
w:st=3D"on">rg</st1:PersonName><br>
Subject: Re: [BEHAVE] Re: ICE deployment data before LC for =
RFC<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><st1:PersonName w:st=3D"on"><font size=3D2 =
face=3D"Courier New"><span
 style=3D'font-size:10.0pt'>Henry =
Sinnreich</span></font></st1:PersonName> skrev:<o:p></o:p></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Following some discussions and soul searching, the best =
approach
for<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; ICE-17 for LC would be to _go ahead and make it an =
experimental
RFC_.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; This would support the development of interoperable
implementations, the<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; collection of deployment data and also hopefully open =
source code.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Though I have full faith in ICE, following the practices =
that made
the<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; IETF successful take precedent here, I =
believe.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; This is a change from the attached that I wrote on =
7/16/2007.<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; Note: We can only hope the proliferation of SBC in VoIP =
service
provider<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; networks will not make these concerns and ICE for SIP a =
moot
issue, but<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>&gt; this is another topic.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Henry,<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I don't quite understand why you are requesting a status change =
of ICE?
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>To me it appears that ICE is almost finished spec. It has been =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>implemented, used and feedback has been influencing the =
protocol. It is
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>definitly one of the better examples of current IETF work that =
actually
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>has running code. Sure, the implementations may not be fully =
updated
yet <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>to the latest draft version. But I think it is safe to say that =
ICE <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>works in the intended core cases. If there are some corner cases =
where <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>it fails, that may be. But that is not information we will have =
until <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>really la<st1:PersonName w:st=3D"on">rg</st1:PersonName>e scale
deployement or someone makes very extensive =
tests.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>I think the IETF and the Internet has much more benefit from a
standards <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>track solution than any of the risks existing with ICE of =
today.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>And to be clear, proposed standard is allowed to be published =
without a
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>single implementation exists. It might not be good practice, but =
it is <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>allowed by our process. And as I said before ICE has had enough =
<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>implementation that I am very confortable in balloting YES for =
it when <o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>it appears on the IESGs table.<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Cheers<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Magnus Westerlund<o:p></o:p></span></font></p>

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

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>IETF Transport Area Director &amp; TSVWG =
Chair<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----------------------------------------------------------------=
-----<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Multimedia Technologies, Ericsson Research =
EAB/TVM/M<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----------------------------------------------------------------=
-----<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><st1:place w:st=3D"on"><st1:City =
w:st=3D"on"><font size=3D2
  face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>Ericsson</span></font></st1:City>
 <st1:State =
w:st=3D"on">AB</st1:State></st1:place>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| Phone +46 8 4048287<o:p></o:p></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>Torshamsgatan
23&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |
Fax&nbsp;&nbsp; +46 8 7575550<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>S-164 80 Stockholm, Sweden | mailto: =
magnus.westerlund@ericsson.com<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>-----------------------------------------------------------------=
-----<o:p></o:p></span></font></p>

</div>

</body>

</html>

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From sip-bounces@ietf.org Wed Jul 18 11:27:09 2007
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Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
Date: Wed, 18 Jul 2007 10:26:54 -0500
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> > What I mean is that the error response may not be delivered=20
> to the UAC=20
> > because a forking proxy will only deliver one final=20
> response. If there=20
> > are multiple error responses on the individual forks, one being the=20
> > one with the RPH, and the other being some other response,=20
> it is up to=20
> > the proxy to decide which one to forward back.
> >=20
> > So the UAC can not rely on receiving the RPH to "re-intiate" the=20
> > session or provide some alternative treatment. This may or=20
> may not be=20
> > a problem depending on the application (that's why I said "if it=20
> > absolutely has").
> >=20
>=20
> I don't think that will be a problem.=20
>=20
> For one thing (for the namespaces I am working with) the=20
> expectation would be that ALL the responses (except 100 and=20
> 403) would have the RPH on the response, so it wouldn't=20
> matter which one made it back to the UAC.=20

I am not talking about multiple provisional responses from the same
endpoint.

I am talking about multiple responses (provisional and final) from=20
multiple forks. For example, multiple 4XX responses.

While it is possible that a single forking proxy may decide to=20
include the RPH in all responses, that proxy has absolutely no=20
control over downstream and upstream proxies who may not include it
(they may be proxies in different domains).

That's the point I'm trying to make. You can not enforce the=20
presence RPH in responses in a global system.

> Secondly, the expectation is that the devices responsible for=20
> "re-initiating" WOULD be stateful, and not dependant on the=20
> information in the RPH in the response.  It is the=20
> intervening devices that might not be stateful, and would=20
> take advantage of the RPH information in the responses.=20

That is precisely the type of clarfication that needs to be included
in the draft. It points out the constrainsts of the mechanism so=20
implementors use the feature properly.

Some text explaining that the UAC needs to be able to deal with the RPH
not being present in responses gracefully (and why) is needed.

Similar text for proxies also need to be there. What if a proxy=20
forwards a request with RPH and it receives nultiple responses from
the downstream proxies, with some including the RPH and others not
including it? Does this need to be standardized?


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From sip-bounces@ietf.org Wed Jul 18 12:13:05 2007
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:

> Henry Sinnreich skrev:
>> Following some discussions and soul searching, the best approach for
>> ICE-17 for LC would be to _go ahead and make it an experimental RFC_.
>> This would support the development of interoperable  
>> implementations, the
>> collection of deployment data and also hopefully open source code.
>> Though I have full faith in ICE, following the practices that made  
>> the
>> IETF successful take precedent here, I believe.
>> This is a change from the attached that I wrote on 7/16/2007.
>> Note: We can only hope the proliferation of SBC in VoIP service  
>> provider
>> networks will not make these concerns and ICE for SIP a moot  
>> issue, but
>> this is another topic.
>
> Henry,
>
> I don't quite understand why you are requesting a status change of  
> ICE? To me it appears that ICE is almost finished spec. It has been  
> implemented, used and feedback has been influencing the protocol.  
> It is definitly one of the better examples of current IETF work  
> that actually has running code. Sure, the implementations may not  
> be fully updated yet to the latest draft version. But I think it is  
> safe to say that ICE works in the intended core cases. If there are  
> some corner cases where it fails, that may be. But that is not  
> information we will have until really large scale deployement or  
> someone makes very extensive tests.

Henry has, I believe, been very consistent in urging the IETF to have  
more "running code" experience on every specification (not just ICE)  
than our process requires. He has, IIRC, also advocated changing the  
process to require running code before publishing an RFC. He's even  
suggested at times that one should have running code before  
submitting a protocol draft.

Another point to consider -- I've known Henry to bark at a tree with  
no squirrel in it occasionally, but most of the time he's pretty  
reliable, more so than most technology predictors. If he's worried  
about ICE, it's probably because he's been talking to people who are  
trying to implement it and they're telling him it is difficult or  
that they're having performance or reliability issues. I imagine that  
there are plenty of internal development teams in various companies  
that are working on this and not necessarily bringing their results  
back to the IETF. After all, if everybody sees the emperor as fully  
clothed, they'd be embarrassed to proclaim otherwise. Of course, I  
wish all of these developers were tied into the IETF, but then I'm  
sure I don't want to see every ice-implementor's question on the SIP  
list. That is why we have a sip-implementors mailing list, right?

Personally, I think ICE is the best solution we have. We KNOW of  
cases where hole-punching doesn't work, and I think ICE will produce  
usable results in most of those cases. But that's just my personal  
opinion from reading the specs. I haven't implemented ICE, and I  
haven't used anybody else's implementation. Sure, it's complex, but  
that's because the underlying problem is complex -- there are just  
too many different behaviors out there in "NAT and firewall land". Of  
course, I could be wrong, too. It's happened before, once or  
twice ;-). Just how sure are we that we're right about this?

Are we designing a "spruce goose" or a practical system? Only  
experience will tell us. But some good flight reports from the  
prototypes could go a long way towards reassuring us of the design's  
utility.

So far we've had a couple of people report hearsay evidence -- that  
somebody else has an ICE implementation and it works. Any of you  
folks have personal experience you can share? Have there been  
prototypes exercised at the interop events?

--
Dean




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For those of you who like agendas in email:

SIP Agenda IETF 69
]
Session 1, Monday July 23 0900-1130 Red Lacquer
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0915
SIPS WGLC
Francois Audet
draft-ietf-sip-sips-05.txt
0935
Outbound WGLC
Rohan Mahy
Cullen Jennings
draft-ietf-sip-outbound-10.txt
1005
Delivering R-URI and Parameters to UA
Jonathan Rosenberg
draft-rosenberg-sip-ua-loose-route-01.txt
1040
Resource Priority Header Issues
James Polk
draft-polk-sip-rph-in-responses-00
draft-polk-sip-rph-new-namespaces-01.txt

1105
MIME Body Handling
Gonzalo Camarillo
draft-camarillo-sip-body-handling-01.txt
1130
End of Session



Session 2, Tuesday July 24 0900-1130 Grand Ballroom
Start Time
Topic
Discussion Lead
Reading List
0900
Agenda Bash and Status
Chairs
This document
0905
Fork Loop Fix and Corrections
Robert Sparks
draft-ietf-sip-fork-loop-fix-05
draft-sparks-sipping-max-breadth-01
draft-drage-sip-essential-correction-01

0925
SAML
Hannes Tschofenig
Jeff Hodges
draft-ietf-sip-saml-02
0945
eTags For Notification
Aki Niemi
draft-ietf-sip-subnot-etags-00.txt
1000
UA-Driven Privacy
Mayumi Munakata
draft-munakata-sip-privacy-new-01.txt
1020
Domain Certs
Vijay Gurbani
Scott Lawrence
draft-gurbani-sip-domain-certs-06.txt
1045
Certificate Authentication
Steve Dotson
draft-dotson-sip-certificate-auth-03.txt
1100
INFO Considered Harmful
Eric Burger
draft-burger-sip-info-00
1115
Media identity
Dan Wing
draft-wing-sip-identity-media-00.txt
1130
End of session




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 style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 2px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"6"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
23.9px;">SIP Agenda IETF 69</SPAN></FONT></P><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">]</SPAN></FONT></DIV><P style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 5px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"5"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 16.8px;"><B>Session 1, =
Monday July 23 0900-1130 Red Lacquer</B></SPAN></FONT></P><TABLE =
cellspacing=3D"0" cellpadding=3D"0" style=3D"background-color: #ffffff; =
border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 2.0px 0.0px =
2.0px 0.0px"><TBODY><TR><TD valign=3D"middle" style=3D"width: 64.0px; =
margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 0.0px 2.0px 0.0px 2.0px"><DIV style=3D"text-align: =
center;margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><B>Start Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0915</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">SIPS WGLC</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Francois Audet</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-05.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-sips-05.txt</SPAN></FONT></A></DIV></TD></TR><TR><=
TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0935</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Outbound WGLC</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Rohan Mahy</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Cullen Jennings</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-10.txt=
"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-outbound-10.txt</SPAN></FONT></A></DIV></TD></TR><=
TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1005</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Delivering R-URI and Parameters to =
UA</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Jonathan Rosenberg</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-rosenberg-sip-ua-loose-r=
oute-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-rosenberg-sip-ua-loose-route-01.txt</SPAN></FONT></A></DIV>=
</TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: =
0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px =
1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; padding: =
0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">1040</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Resource Priority Header Issues</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">James Polk</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-response=
s-00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-in-responses-00</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespa=
ces-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-polk-sip-rph-new-namespaces-01.txt</SPAN></FONT></A></P></T=
D></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1105</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">MIME Body =
Handling</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
119.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Gonzalo Camarillo</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handl=
ing-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-camarillo-sip-body-handling-01.txt</SPAN></FONT></A></DIV><=
/TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1130</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 229.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">End of Session</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 119.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12.7px/normal Helvetica; min-height: 16px; "><SPAN =
class=3D"Apple-style-span" style=3D"line-height: =
19px;"><BR></SPAN></DIV></TD><TD valign=3D"middle" style=3D"width: =
252.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12.7px/normal Helvetica; =
min-height: 16px; "><SPAN class=3D"Apple-style-span" style=3D"line-height:=
 19px;"><BR></SPAN></DIV></TD></TR></TBODY></TABLE><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><BR class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><P =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 5px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"5"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
16.8px;"><B>Session 2, Tuesday July 24 0900-1130 Grand =
Ballroom</B></SPAN></FONT></P><TABLE cellspacing=3D"0" cellpadding=3D"0" =
style=3D"background-color: #ffffff; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 2.0px 0.0px 2.0px 0.0px"><TBODY><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Start =
Time</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Topic</B></SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"text-align: center;margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;"><B>Discussion =
Lead</B></SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"text-align: center;margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;"><B>Reading =
List</B></SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0900</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Agenda Bash and Status</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Chairs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">This =
document</SPAN></FONT></DIV></TD></TR><TR><TD valign=3D"middle" =
style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0905</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Fork Loop Fix and Corrections</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Robert Sparks</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-fork-loop-fix-0=
5.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-fork-loop-fix-05</SPAN></FONT></A><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">=A0</SPAN></FONT></DIV><P style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 6px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sipping-max-bread=
th-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-sparks-sipping-max-breadth-01</SPAN></FONT></A><SPAN =
class=3D"Apple-style-span" style=3D"line-height: 19px;"><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px; line-height: =
normal;"> </SPAN></FONT><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px; line-height: =
normal;"><BR></SPAN></FONT></SPAN><A =
href=3D"http://www.ietf.org/internet-drafts/draft-drage-sip-essential-corr=
ection-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;"><FONT class=3D"Apple-style-span" =
color=3D"#2367BD">draft-drage-sip-essential-correction-01</FONT></SPAN></F=
ONT></A></P></TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; =
margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; border-width: =
1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 #808080 #808080; =
padding: 0.0px 2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">0925</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">SAML</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><SPAN class=3D"Apple-style-span" style=3D"line-height:=
 19px;"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: 12.7px; =
line-height: normal;">Hannes Tschofenig</SPAN></FONT><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px; line-height: =
normal;">=A0</SPAN></FONT></SPAN></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 12.7px;">Jeff =
Hodges</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-saml-02.txt"><F=
ONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-saml-02</SPAN></FONT></A></DIV></TD></TR><TR><TD =
valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">0945</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">eTags For =
Notification</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width:=
 116.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Aki Niemi</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00=
.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN=
 class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-ietf-sip-subnot-etags-00.txt</SPAN></FONT></A></DIV></TD></=
TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px =
0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1000</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">UA-Driven =
Privacy</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
116.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Mayumi Munakata</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new=
-01.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-munakata-sip-privacy-new-01.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1020</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Domain Certs</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Vijay Gurbani</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Scott Lawrence</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs=
-06.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-gurbani-sip-domain-certs-06.txt</SPAN></FONT></A></DIV></TD=
></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1045</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Certificate =
Authentication</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Steve Dotson</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-a=
uth-03.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-dotson-sip-certificate-auth-03.txt</SPAN></FONT></A></DIV><=
/TD></TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1100</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">INFO Considered =
Harmful</SPAN></FONT></DIV></TD><TD valign=3D"middle" style=3D"width: =
116.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: solid; =
border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 #808080 =
#808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">Eric Burger</SPAN></FONT></DIV></TD><TD valign=3D"middle" =
style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px 0.5px; border-style: =
solid; border-width: 1.0px 1.0px 1.0px 1.0px; border-color: #808080 =
#808080 #808080 #808080; padding: 0.0px 2.0px 0.0px 2.0px"><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-burger-sip-info-00"><FON=
T class=3D"Apple-style-span" face=3D"Helvetica" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-burger-sip-info-00</SPAN></FONT></A></DIV></TD></TR><TR><TD=
 valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1115</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Media identity</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">Dan Wing</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 231.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><A =
href=3D"http://www.ietf.org/internet-drafts/draft-wing-sip-identity-media-=
00.txt"><FONT class=3D"Apple-style-span" face=3D"Helvetica" =
size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
12.7px;">draft-wing-sip-identity-media-00.txt</SPAN></FONT></A></DIV></TD>=
</TR><TR><TD valign=3D"middle" style=3D"width: 64.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">1130</SPAN></FONT></DIV></TD><TD =
valign=3D"middle" style=3D"width: 175.0px; margin: 0.0px 0.5px 0.0px =
0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px 1.0px; =
border-color: #808080 #808080 #808080 #808080; padding: 0.0px 2.0px =
0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><FONT class=3D"Apple-style-span" =
face=3D"Helvetica" size=3D"3"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 12.7px;">End of session</SPAN></FONT></DIV></TD><TD =
colspan=3D"2" valign=3D"middle" style=3D"width: 116.0px; margin: 0.0px =
0.5px 0.0px 0.5px; border-style: solid; border-width: 1.0px 1.0px 1.0px =
1.0px; border-color: #808080 #808080 #808080 #808080; padding: 0.0px =
2.0px 0.0px 2.0px"><DIV style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12.7px/normal Helvetica; min-height: 16px; "><SPAN =
class=3D"Apple-style-span" style=3D"line-height: =
19px;"><BR></SPAN></DIV></TD></TR></TBODY></TABLE><DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV><BR =
class=3D"khtml-block-placeholder"></DIV></BODY></HTML>=

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From sip-bounces@ietf.org Wed Jul 18 12:27:02 2007
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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
Date: Wed, 18 Jul 2007 12:26:55 -0400
To: Pyda Srisuresh <srisuresh@yahoo.com>
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Frank, Henry, Suresh:

I agree that ICE is a complex spec. I also agree that it would be =20
very interesting to see
more data on which situations result in a direct path between =20
endpoints, as opposed to
a path via a TURN server.

However, I don't understand the comment that "Hole Punching =20
Techniques" are an alternative.
In my view, ICE _is_ a "Hole Punching Technique".

Perhaps you are referring to the use of STUN alone???
The problem with using STUN alone is that there are cases where using =20=

the reflexive address
doesn't work, and there is no way with STUN alone to determine when =20
an endpoint should or
should not use a reflexive address.

But perhaps you are referring to something else?
- Philip


On 18-Jul-07, at 11:37 , Pyda Srisuresh wrote:

> Thanks to Henry and Frank for so eloquently articulating this.
>
> As both Henry and Frank mentioned, I fully support standardizing on =20=

> HOLE
> PUNCHING TECHNIQUE and leveraging the IETF=92s considerable influence =20=

> to force
> NAT vendors to migrate towards implementations that are friendly to =20=

> Hole
> Punching and the Application developers to design applications =20
> using the hole
> punching techniques. That should be the FIRST order of priority.
>
> Yet, what is happening in BEHAVE WG is the other way around. Some =20
> people have
> put their weight so heavily in favor of ICE that they asked to drop =20=

> the work on
> hole punching technique as WG item. THIS IS WRONG. I no longer see =20
> "NAT
> Traveral application design guidelines" as BEHAVE WG milestone.
>
> Magnus - Can you explain why the IETF is not allowing HOLE PUNCHING =20=

> TECHNIQUES
> to move forward as a WG item?
>
> regards,
> suresh
>
> --- "Frank W. Miller" <fwmiller@cornfed.com> wrote:
>
>>
>>
>> Greetings,
>>
>>
>>
>> What I think Henry is advocating is a systematic approach to =20
>> making sure
>> that the content of the ICE spec is right for its intended =20
>> purpose, i.e. a
>> complete, general NAT traversal solution.  NAT traversal has been =20
>> in my
>> opinion one of the most difficult aspects of SIP implementation =20
>> and its very
>> important that its eventual standardized solution be right.  I =20
>> know I spent
>> a lot of time implementing all the "tests" in the first 3489, only to
>> discover that they were not that useful.  There are two valid =20
>> technical
>> concerns that are put forth in this email:
>>
>>
>>
>> 1)     Viability of correct operation in practical =20
>> implementations.  ICE can
>> result in very complicated message transfers.
>>
>> 2)     Performance.  As pointed out in this email and in other =20
>> places, the
>> additional latency associated with ICE operations can be significant.
>>
>>
>>
>> Henry also points out that there are competing approaches (e.g. Hole
>> Punching) that can perform well in many situations and have better
>> performance.  IMHO, another approach by the IETF in general would =20
>> be to
>> standardize on Hole Punching and leverage the IETF's considerable =20
>> influence
>> to force NAT vendors to migrate towards implementations that are =20
>> friendly to
>> Hole Punching.
>>
>>
>>
>> The bottom line is that I agree with Henry in that labeling ICE as an
>> experimental draft is the right thing to do.  It will allow those
>> implementing ICE (including myself in all likelihood) to proceed =20
>> but signals
>> to the users the correct status of the standard's contents.
>>
>>
>>
>> I also want to applaud loudly the underlying tone of Henry's =20
>> thread here.
>> That is, in solving specific technical problems within the IETF =20
>> forum,
>> focusing on empirical observation of working implementations =20
>> should be the
>> MO that we all strive for when developing and proposing what may =20
>> become
>> protocol standards.
>>
>>
>>
>> Thanks,
>>
>> FM
>>
>>
>>
>>
>>
>>   _____
>>
>> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
>> Sent: Wednesday, July 18, 2007 8:20 AM
>> To: Magnus Westerlund
>> Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
>> Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>>
>>
>> Magnus,
>>
>>
>>
>> I sincerely appreciate your interest and this note!
>>
>>
>>
>>> I don't quite understand why you are requesting a status change =20
>>> of ICE
>>
>>
>>
>> However the arguments stated by the WG co-chairs and by you here are
>> procedural in nature, whereas the facts are some cause for serious =20=

>> concern,
>> and I will just mention a few facts:
>>
>>
>>
>> RFC 3489 (STUN) was also presented as a NAT traversal solution =20
>> back in 2003,
>> but <draft-ietf-behave-rfc3489bis-07> now clearly admits that =20
>> "STUN is not a
>> NAT traversal solution by itself. Rather, it is a tool to be used =20
>> in the
>> context of a NAT traversal solution. This is an important change =20
>> from the
>> previous version of this specification (RFC 3489), which presented =20=

>> STUN as a
>> complete solution."
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-behave-=20
>> rfc3489bis-07.txt
>>
>> My guess is, in plain English, RFC 3489 turned out just NOT to be an
>> acceptable solution as it was hoped to be.
>>
>>
>>
>> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the =20
>> ICE call
>> flow with two endpoints behind two NAT (Fig. 22 on page 38) =20
>> require 66
>> messages just for the SIP INVITE!
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-=20
>> scenarios-07.txt
>>
>> What happens if the two endpoints are in different ISPs that have =20
>> NAT of
>> their own in addition to the ones in Fig. 22?
>>
>> What happens if one would try ICE along a P2PSIP route with say =20
>> 5-8 hops?
>> Would there be hundreds of messages? What is the likelihood for =20
>> failure?
>>
>>
>>
>> By contrast with such approaches based on hope and faith, the =20
>> BEHAVE WG has
>> produced work based on real measurements and tools to do the =20
>> measurements,
>> such as
>>
>>
>>
>> "NAT Classification Test Results"
>>
>>
>>
>> "Application Design Guidelines for Traversal through Network Address
>> Translators"
>>
>> The measurements for hole punching are reported in
>>
>> "Peer-to-Peer Communication Across Network Address Translators"
>>
>> http://www.brynosaurus.com/pub/net/p2pnat/
>>
>>
>>
>> I am also a believer in ICE and have actually urged Jonathan =20
>> Rosenberg to do
>> this work and he has given his best. Now that we have ICE-17 =20
>> version, it is
>> in the best interest of the industry and the IETF to make it an =20
>> experimental
>> RFC, but not yet a standard so that:
>>
>>  1. Independent, compliant implementations will be developed,
>>
>>  2. Testing in events such as SIPit,
>>
>>  3. Reporting the results, such as reported for the "hole =20
>> punching" above.
>>
>>
>>
>> What happens if indeed hole punching proves to be the far more =20
>> effective
>> approach?
>>
>>
>>
>> The industry can ill afford a new series of RFC[ICE]-bis and all =20
>> the years
>> it takes to move from believing to measuring and proving. I am =20
>> sure IETF
>> procedures were meant in this spirit (though I am not good in =20
>> producing the
>> right quotes).
>>
>>
>>
>> Thanks again for your attention to this concern about ICE.
>>
>> Also my apology to all the anguish to the numerous contributors to =20=

>> the ICE
>> related I-Ds, but writing test tools and reporting measurements is =20=

>> the right
>> way to go.
>>
>>
>>
>> Henry
>>
>>
>>
>>
>>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Wednesday, July 18, 2007 1:11 AM
>> To: Henry Sinnreich
>> Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
>> Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>>
>>
>> Henry Sinnreich skrev:
>>
>>> Following some discussions and soul searching, the best approach for
>>
>>> ICE-17 for LC would be to _go ahead and make it an experimental =20
>>> RFC_.
>>
>>>
>>
>>> This would support the development of interoperable =20
>>> implementations, the
>>
>>> collection of deployment data and also hopefully open source code.
>>
>>>
>>
>>> Though I have full faith in ICE, following the practices that =20
>>> made the
>>
>>> IETF successful take precedent here, I believe.
>>
>>>
>>
>>> This is a change from the attached that I wrote on 7/16/2007.
>>
>>>
>>
>>> Note: We can only hope the proliferation of SBC in VoIP service =20
>>> provider
>>
>>> networks will not make these concerns and ICE for SIP a moot =20
>>> issue, but
>>
>>> this is another topic.
>>
>>
>>
>> Henry,
>>
>>
>>
>> I don't quite understand why you are requesting a status change of =20=

>> ICE?
>>
>> To me it appears that ICE is almost finished spec. It has been
>>
>> implemented, used and feedback has been influencing the protocol. =20
>> It is
>>
>> definitly one of the better examples of current IETF work that =20
>> actually
>>
>> has running code. Sure, the implementations may not be fully =20
>> updated yet
>>
>> to the latest draft version. But I think it is safe to say that ICE
>>
>> works in the intended core cases. If there are some corner cases =20
>> where
>>
>> it fails, that may be. But that is not information we will have until
>>
>> really large scale deployement or someone makes very extensive tests.
>>
>>
>>
>> I think the IETF and the Internet has much more benefit from a =20
>> standards
>>
>> track solution than any of the risks existing with ICE of today.
>>
>>
>>
>> And to be clear, proposed standard is allowed to be published =20
>> without a
>>
>> single implementation exists. It might not be good practice, but =20
>> it is
>>
>> allowed by our process. And as I said before ICE has had enough
>>
>> implementation that I am very confortable in balloting YES for it =20
>> when
>>
>> it appears on the IESGs table.
>>
>>
>>
>> Cheers
>>
>>
>>
>> Magnus Westerlund
>>
>>
>>
>> IETF Transport Area Director & TSVWG Chair
>>
>> ---------------------------------------------------------------------=20=

>> -
>>
>> Multimedia Technologies, Ericsson Research EAB/TVM/M
>>
>> ---------------------------------------------------------------------=20=

>> -
>>
>> Ericsson AB                | Phone +46 8 4048287
>>
>> Torshamsgatan 23           | Fax   +46 8 7575550
>>
>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>> ---------------------------------------------------------------------=20=

>> -
>>
>>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
>>
>
>
>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave



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In line.

"Francois Audet" <audet@nortel.com> wrote on 07/18/2007 11:26:54 AM:

> > > What I mean is that the error response may not be delivered 
> > to the UAC 
> > > because a forking proxy will only deliver one final 
> > response. If there 
> > > are multiple error responses on the individual forks, one being the 
> > > one with the RPH, and the other being some other response, 
> > it is up to 
> > > the proxy to decide which one to forward back.
> > > 
> > > So the UAC can not rely on receiving the RPH to "re-intiate" the 
> > > session or provide some alternative treatment. This may or 
> > may not be 
> > > a problem depending on the application (that's why I said "if it 
> > > absolutely has").
> > > 
> > 
> > I don't think that will be a problem. 
> > 
> > For one thing (for the namespaces I am working with) the 
> > expectation would be that ALL the responses (except 100 and 
> > 403) would have the RPH on the response, so it wouldn't 
> > matter which one made it back to the UAC. 
> 
> I am not talking about multiple provisional responses from the same
> endpoint.
> 
> I am talking about multiple responses (provisional and final) from 
> multiple forks. For example, multiple 4XX responses.

Yes, I understood that.
> 
> While it is possible that a single forking proxy may decide to 
> include the RPH in all responses, that proxy has absolutely no 
> control over downstream and upstream proxies who may not include it
> (they may be proxies in different domains).
> 
> That's the point I'm trying to make. You can not enforce the 
> presence RPH in responses in a global system.

Agree. For the context I am working in, a "global system" is not of 
concern.  I would not expect universal "RPH in responses" in a global 
system.

> 
> > Secondly, the expectation is that the devices responsible for 
> > "re-initiating" WOULD be stateful, and not dependant on the 
> > information in the RPH in the response.  It is the 
> > intervening devices that might not be stateful, and would 
> > take advantage of the RPH information in the responses. 
> 
> That is precisely the type of clarification that needs to be included
> in the draft. It points out the constraints of the mechanism so 
> implementors use the feature properly.
> 
> Some text explaining that the UAC needs to be able to deal with the RPH
> not being present in responses gracefully (and why) is needed.

I am not sure why that needs to be in the ID.  It is an implementation 
specific issue. 

Since  RFC 4412, as currently published, forbids RPH in responses, I think 
it "goes without saying" that "the UAC needs to be able to deal with the 
RPH not being present in responses".

Further, while I know that the systems (and namespaces) _I_ work with 
intend to use RPH in responses, I also have been told that other systems 
(using other namespaces) do not intend to put RPH in ANY responses.  I do 
not think there is any need to standardize this aspect, just to adjust the 
standard so it is PERMITTED.
> 
> Similar text for proxies also need to be there. What if a proxy 
> forwards a request with RPH and it receives nultiple responses from
> the downstream proxies, with some including the RPH and others not
> including it? Does this need to be standardized?

Again, I do not think there is a need to standardize this aspect.

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

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<br><font size=2 face="sans-serif"><br>
In line.</font>
<br>
<br><font size=2><tt>&quot;Francois Audet&quot; &lt;audet@nortel.com&gt;
wrote on 07/18/2007 11:26:54 AM:<br>
<br>
&gt; &gt; &gt; What I mean is that the error response may not be delivered
<br>
&gt; &gt; to the UAC <br>
&gt; &gt; &gt; because a forking proxy will only deliver one final <br>
&gt; &gt; response. If there <br>
&gt; &gt; &gt; are multiple error responses on the individual forks, one
being the <br>
&gt; &gt; &gt; one with the RPH, and the other being some other response,
<br>
&gt; &gt; it is up to <br>
&gt; &gt; &gt; the proxy to decide which one to forward back.<br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; So the UAC can not rely on receiving the RPH to &quot;re-intiate&quot;
the <br>
&gt; &gt; &gt; session or provide some alternative treatment. This may
or <br>
&gt; &gt; may not be <br>
&gt; &gt; &gt; a problem depending on the application (that's why I said
&quot;if it <br>
&gt; &gt; &gt; absolutely has&quot;).<br>
&gt; &gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; I don't think that will be a problem. <br>
&gt; &gt; <br>
&gt; &gt; For one thing (for the namespaces I am working with) the <br>
&gt; &gt; expectation would be that ALL the responses (except 100 and <br>
&gt; &gt; 403) would have the RPH on the response, so it wouldn't <br>
&gt; &gt; matter which one made it back to the UAC. <br>
&gt; <br>
&gt; I am not talking about multiple provisional responses from the same<br>
&gt; endpoint.<br>
&gt; <br>
&gt; I am talking about multiple responses (provisional and final) from
<br>
&gt; multiple forks. For example, multiple 4XX responses.</tt></font>
<br>
<br><font size=2><tt>Yes, I understood that.<br>
&gt; <br>
&gt; While it is possible that a single forking proxy may decide to <br>
&gt; include the RPH in all responses, that proxy has absolutely no <br>
&gt; control over downstream and upstream proxies who may not include it<br>
&gt; (they may be proxies in different domains).<br>
&gt; <br>
&gt; That's the point I'm trying to make. You can not enforce the <br>
&gt; presence RPH in responses in a global system.</tt></font>
<br>
<br><font size=2><tt>Agree. For the context I am working in, a &quot;global
system&quot; is not of concern. &nbsp;I would not expect universal &quot;RPH
in responses&quot; in a global system.</tt></font>
<br><font size=2><tt><br>
&gt; <br>
&gt; &gt; Secondly, the expectation is that the devices responsible for
<br>
&gt; &gt; &quot;re-initiating&quot; WOULD be stateful, and not dependant
on the <br>
&gt; &gt; information in the RPH in the response. &nbsp;It is the <br>
&gt; &gt; intervening devices that might not be stateful, and would <br>
&gt; &gt; take advantage of the RPH information in the responses. <br>
&gt; <br>
&gt; That is precisely the type of clarification that needs to be included<br>
&gt; in the draft. It points out the constraints of the mechanism so <br>
&gt; implementors use the feature properly.<br>
&gt; <br>
&gt; Some text explaining that the UAC needs to be able to deal with the
RPH<br>
&gt; not being present in responses gracefully (and why) is needed.</tt></font>
<br>
<br><font size=2><tt>I am not sure why that needs to be in the ID. &nbsp;It
is an implementation specific issue. &nbsp;</tt></font>
<br>
<br><font size=2><tt>Since &nbsp;RFC 4412, as currently published, forbids
RPH in responses, I think it &quot;goes without saying&quot; that &quot;the
UAC needs to be able to deal with the RPH not being present in responses&quot;.</tt></font>
<br>
<br><font size=2><tt>Further, while I know that the systems (and namespaces)
_I_ work with intend to use RPH in responses, I also have been told that
other systems (using other namespaces) do not intend to put RPH in ANY
responses. &nbsp;I do not think there is any need to standardize this aspect,
just to adjust the standard so it is PERMITTED.<br>
&gt; <br>
&gt; Similar text for proxies also need to be there. What if a proxy <br>
&gt; forwards a request with RPH and it receives nultiple responses from<br>
&gt; the downstream proxies, with some including the RPH and others not<br>
&gt; including it? Does this need to be standardized?</tt></font>
<br>
<br><font size=2><tt>Again, I do not think there is a need to standardize
this aspect.</tt></font>
<br>
<br><font size=2><tt>Janet<br>
&gt; <br>
&gt; <br>
&gt; _______________________________________________<br>
&gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
&gt; Use sipping@ietf.org for new developments on the application of sip<br>
</tt></font>
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> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows 
> the ICE call flow with two endpoints behind two NAT (Fig. 22 
> on page 38) require 66 messages just for the SIP INVITE!
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-sce
> narios-07.txt 

That figure does not show 66 messages.

That figure is clearly showing how the STUN messages are NATted,
which leads to an incorrect conclusion around the number of 
messages.  Specifically, for example, it shows this message
exchange which might be counted as "four messages"; yet, there
are only two messages (the allocate request and the allocate
response), which are NATted:

   L               NAT             STUN              
                                   Server
   |                |                |               
   |(1) Alloc Req   |                |               
   |Src=L-PRIV-1    |                |               
   |Dest=STUN-PUB-1 |                |               
   |--------------->|                |               
   |                |                |               
   |                |(2) Alloc Req   |               
   |                |Src=L-NAT-PUB-1 |               
   |                |Des=STUN-PUB-1  |               
   |                |--------------->|               
   |                |                |               
   |                |(3) Alloc Resp  |               
   |                |<---------------|               
   |                |Src=STUN-PUB-1  |               
   |                |Dest=L-NAT-PUB-1|               
   |                |Map=L-NAT-PUB-1 |               
   |                |Rel=STUN-PUB-2  |               
   |                |                |               
   |(4) Alloc Resp  |                |               
   |<---------------|                |               
   |Src=STUN-PUB-1  |                |               
   |Dest=L-PRIV-1   |                |               
   |Map=L-NAT-PUB-1 |                |               
   |Rel=STUN-PUB-2  |                |               
   |                |                |               

If we added another NAT on that path, the number of messages
on the network does not double.  The number of messages on 
the network stays the same -- the messages merely have their
IP and UDP headers rewritten by the NATs.  That NAT function
does not increase the number of messages on the network.



Similarly, another message shown in that same figure is also
one message; however, it's shown as two messages:

L               NAT             STUN              NAT               R
                                Server
|(9) SIP INVITE  |                |                |                |
|------------------------------------------------->|                |
|                |                |                |(10) SIP INVITE |
|                |                |                |--------------->|

As you know, there aren't two messages -- it's one message that is
NATted.

If message 9 and 10 were shown as clearly as messages 1-4, messages
9-10 would be shown as four messages (one from L to its NAT, another 
from NAT to NAT, and the forth (shown) from NAT to R).  But this is
just one SIP message, which is NATted.  It is inaccurate  to count
it as two separate messages or as four separate messages, because
it is only one packet that is NATted.  Counting a STUN packet,
or a DNS packet, as a separate message each time it traverses a 
NAT is similarly inaccurate.

By my count, figure 22 exchanges 24 STUN/ICE messages and 6 
SIP messages.

> What happens if the two endpoints are in different ISPs that 
> have NAT of their own in addition to the ones in Fig. 22?

The same 24 STUN/ICE messages would be exchanged, and the same
6 SIP messages would be exchanged.

> What happens if one would try ICE along a P2PSIP route with 
> say 5-8 hops?

The SIP signaling messages would traverse those 5-8 additional 
hops between the P2PSIP nodes participating in the SIP proxy
overlay.  The STUN/ICE messages are sent peer-to-peer, and 
would remain the same 24 messages if you're doing the same 
Figure 22 flow.  

> Would there be hundreds of messages?

No.

-d


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Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
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To: "Janet P Gunn" <jgunn6@csc.com>
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I will be happy if the document is revised to explain that you cannot
expect that responses received by a UAC or Proxy will necessarily
include RPM, in particular in systems where forking occurs, and that
therefore, applications that use this draft must take it into account.


________________________________

	From: Janet P Gunn [mailto:jgunn6@csc.com]=20
	Sent: Wednesday, July 18, 2007 10:41
	To: Audet, Francois (SC100:3055)
	Cc: sip@ietf.org; James M. Polk
	Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
=09
=09

=09
	In line.=20
=09
	"Francois Audet" <audet@nortel.com> wrote on 07/18/2007 11:26:54
AM:
=09
	> > > What I mean is that the error response may not be
delivered=20
	> > to the UAC=20
	> > > because a forking proxy will only deliver one final=20
	> > response. If there=20
	> > > are multiple error responses on the individual forks, one
being the=20
	> > > one with the RPH, and the other being some other response,

	> > it is up to=20
	> > > the proxy to decide which one to forward back.
	> > >=20
	> > > So the UAC can not rely on receiving the RPH to
"re-intiate" the=20
	> > > session or provide some alternative treatment. This may or

	> > may not be=20
	> > > a problem depending on the application (that's why I said
"if it=20
	> > > absolutely has").
	> > >=20
	> >=20
	> > I don't think that will be a problem.=20
	> >=20
	> > For one thing (for the namespaces I am working with) the=20
	> > expectation would be that ALL the responses (except 100 and=20
	> > 403) would have the RPH on the response, so it wouldn't=20
	> > matter which one made it back to the UAC.=20
	>=20
	> I am not talking about multiple provisional responses from the
same
	> endpoint.
	>=20
	> I am talking about multiple responses (provisional and final)
from=20
	> multiple forks. For example, multiple 4XX responses.=20
=09
	Yes, I understood that.
	>=20
	> While it is possible that a single forking proxy may decide to

	> include the RPH in all responses, that proxy has absolutely no

	> control over downstream and upstream proxies who may not
include it
	> (they may be proxies in different domains).
	>=20
	> That's the point I'm trying to make. You can not enforce the=20
	> presence RPH in responses in a global system.=20
=09
	Agree. For the context I am working in, a "global system" is not
of concern.  I would not expect universal "RPH in responses" in a global
system.=20
=09
	>=20
	> > Secondly, the expectation is that the devices responsible
for=20
	> > "re-initiating" WOULD be stateful, and not dependant on the=20
	> > information in the RPH in the response.  It is the=20
	> > intervening devices that might not be stateful, and would=20
	> > take advantage of the RPH information in the responses.=20
	>=20
	> That is precisely the type of clarification that needs to be
included
	> in the draft. It points out the constraints of the mechanism
so=20
	> implementors use the feature properly.
	>=20
	> Some text explaining that the UAC needs to be able to deal
with the RPH
	> not being present in responses gracefully (and why) is needed.

=09
	I am not sure why that needs to be in the ID.  It is an
implementation specific issue.  =20
=09
	Since  RFC 4412, as currently published, forbids RPH in
responses, I think it "goes without saying" that "the UAC needs to be
able to deal with the RPH not being present in responses".=20
=09
	Further, while I know that the systems (and namespaces) _I_ work
with intend to use RPH in responses, I also have been told that other
systems (using other namespaces) do not intend to put RPH in ANY
responses.  I do not think there is any need to standardize this aspect,
just to adjust the standard so it is PERMITTED.
	>=20
	> Similar text for proxies also need to be there. What if a
proxy=20
	> forwards a request with RPH and it receives nultiple responses
from
	> the downstream proxies, with some including the RPH and others
not
	> including it? Does this need to be standardized?=20
=09
	Again, I do not think there is a need to standardize this
aspect.=20
=09
	Janet
	>=20
	>=20
	> _______________________________________________
	> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
	> This list is for NEW development of the core SIP Protocol
	> Use sip-implementors@cs.columbia.edu for questions on current
sip
	> Use sipping@ietf.org for new developments on the application
of sip
=09


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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D416025718-18072007><FONT =
face=3DArial=20
color=3D#800000 size=3D2>I will be happy if the document is revised to =
explain that=20
you cannot expect that responses received by a UAC or Proxy&nbsp;will=20
necessarily include RPM, in particular in systems where forking occurs, =
and that=20
therefore, applications that use this draft must take it into=20
account.</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 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> Janet P Gunn =
[mailto:jgunn6@csc.com]=20
  <BR><B>Sent:</B> Wednesday, July 18, 2007 10:41<BR><B>To:</B> Audet, =
Francois=20
  (SC100:3055)<BR><B>Cc:</B> sip@ietf.org; James M. =
Polk<BR><B>Subject:</B> RE:=20
  [Sip] New draft modifying RFC 4412 for Responses<BR></FONT><BR></DIV>
  <DIV></DIV><BR><FONT face=3Dsans-serif size=3D2><BR>In line.</FONT> =
<BR><BR><FONT=20
  size=3D2><TT>"Francois Audet" &lt;audet@nortel.com&gt; wrote on =
07/18/2007=20
  11:26:54 AM:<BR><BR>&gt; &gt; &gt; What I mean is that the error =
response may=20
  not be delivered <BR>&gt; &gt; to the UAC <BR>&gt; &gt; &gt; because a =
forking=20
  proxy will only deliver one final <BR>&gt; &gt; response. If there =
<BR>&gt;=20
  &gt; &gt; are multiple error responses on the individual forks, one =
being the=20
  <BR>&gt; &gt; &gt; one with the RPH, and the other being some other =
response,=20
  <BR>&gt; &gt; it is up to <BR>&gt; &gt; &gt; the proxy to decide which =
one to=20
  forward back.<BR>&gt; &gt; &gt; <BR>&gt; &gt; &gt; So the UAC can not =
rely on=20
  receiving the RPH to "re-intiate" the <BR>&gt; &gt; &gt; session or =
provide=20
  some alternative treatment. This may or <BR>&gt; &gt; may not be =
<BR>&gt; &gt;=20
  &gt; a problem depending on the application (that's why I said "if it =
<BR>&gt;=20
  &gt; &gt; absolutely has").<BR>&gt; &gt; &gt; <BR>&gt; &gt; <BR>&gt; =
&gt; I=20
  don't think that will be a problem. <BR>&gt; &gt; <BR>&gt; &gt; For =
one thing=20
  (for the namespaces I am working with) the <BR>&gt; &gt; expectation =
would be=20
  that ALL the responses (except 100 and <BR>&gt; &gt; 403) would have =
the RPH=20
  on the response, so it wouldn't <BR>&gt; &gt; matter which one made it =
back to=20
  the UAC. <BR>&gt; <BR>&gt; I am not talking about multiple provisional =

  responses from the same<BR>&gt; endpoint.<BR>&gt; <BR>&gt; I am =
talking about=20
  multiple responses (provisional and final) from <BR>&gt; multiple =
forks. For=20
  example, multiple 4XX responses.</TT></FONT> <BR><BR><FONT =
size=3D2><TT>Yes, I=20
  understood that.<BR>&gt; <BR>&gt; While it is possible that a single =
forking=20
  proxy may decide to <BR>&gt; include the RPH in all responses, that =
proxy has=20
  absolutely no <BR>&gt; control over downstream and upstream proxies =
who may=20
  not include it<BR>&gt; (they may be proxies in different =
domains).<BR>&gt;=20
  <BR>&gt; That's the point I'm trying to make. You can not enforce the =
<BR>&gt;=20
  presence RPH in responses in a global system.</TT></FONT> =
<BR><BR><FONT=20
  size=3D2><TT>Agree. For the context I am working in, a "global system" =
is not of=20
  concern. &nbsp;I would not expect universal "RPH in responses" in a =
global=20
  system.</TT></FONT> <BR><FONT size=3D2><TT><BR>&gt; <BR>&gt; &gt; =
Secondly, the=20
  expectation is that the devices responsible for <BR>&gt; &gt; =
"re-initiating"=20
  WOULD be stateful, and not dependant on the <BR>&gt; &gt; information =
in the=20
  RPH in the response. &nbsp;It is the <BR>&gt; &gt; intervening devices =
that=20
  might not be stateful, and would <BR>&gt; &gt; take advantage of the =
RPH=20
  information in the responses. <BR>&gt; <BR>&gt; That is precisely the =
type of=20
  clarification that needs to be included<BR>&gt; in the draft. It =
points out=20
  the constraints of the mechanism so <BR>&gt; implementors use the =
feature=20
  properly.<BR>&gt; <BR>&gt; Some text explaining that the UAC needs to =
be able=20
  to deal with the RPH<BR>&gt; not being present in responses gracefully =
(and=20
  why) is needed.</TT></FONT> <BR><BR><FONT size=3D2><TT>I am not sure =
why that=20
  needs to be in the ID. &nbsp;It is an implementation specific issue.=20
  &nbsp;</TT></FONT> <BR><BR><FONT size=3D2><TT>Since &nbsp;RFC 4412, as =
currently=20
  published, forbids RPH in responses, I think it "goes without saying" =
that=20
  "the UAC needs to be able to deal with the RPH not being present in=20
  responses".</TT></FONT> <BR><BR><FONT size=3D2><TT>Further, while I =
know that=20
  the systems (and namespaces) _I_ work with intend to use RPH in =
responses, I=20
  also have been told that other systems (using other namespaces) do not =
intend=20
  to put RPH in ANY responses. &nbsp;I do not think there is any need to =

  standardize this aspect, just to adjust the standard so it is=20
  PERMITTED.<BR>&gt; <BR>&gt; Similar text for proxies also need to be =
there.=20
  What if a proxy <BR>&gt; forwards a request with RPH and it receives =
nultiple=20
  responses from<BR>&gt; the downstream proxies, with some including the =
RPH and=20
  others not<BR>&gt; including it? Does this need to be=20
  standardized?</TT></FONT> <BR><BR><FONT size=3D2><TT>Again, I do not =
think there=20
  is a need to standardize this aspect.</TT></FONT> <BR><BR><FONT=20
  size=3D2><TT>Janet<BR>&gt; <BR>&gt; <BR>&gt;=20
  _______________________________________________<BR>&gt; Sip mailing =
list=20
  &nbsp;https://www1.ietf.org/mailman/listinfo/sip<BR>&gt; This list is =
for NEW=20
  development of the core SIP Protocol<BR>&gt; Use=20
  sip-implementors@cs.columbia.edu for questions on current sip<BR>&gt; =
Use=20
  sipping@ietf.org for new developments on the application of=20
sip<BR></BLOCKQUOTE></TT></FONT></BODY></HTML>

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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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	"Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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Dan,

What you say makes sense but there is a lingering suspicion that after
admitting that RFC 3489 (STUN), RFC 4091, 4092 (ANAT), etc. just don't
work is that such paper evaluations are in no way a replacement for
proof and measurements. We still need an explanation why the right
approach cannot be taken:

* Publish open source code at Sourceforge.net or similar,
* Publish a test tool as well,
* Gather and publish the results, possibly involving SIPit or another,
* Perfect the OS code and the protocol until it works OK.

This has been done for "Hole Punching for P2P" in=20
http://www.brynosaurus.com/pub/net/p2pnat/=20

What if ICE proves to be no better than its predecessors?=20

So how can one compare something that has tools, measured and published
results with something like ICE that we have to take on pure faith?

My assumption is that hole punching is much more effective for P2PSIP
and I have copied the P2PSIP WG on this. The ratio of messages is
probably 5:1 or higher for ICE vs. hole punching and this is critical
when routing a P2PSIP message over an overlay in 5-10 hops. Of course I
cannot prove it since as yet ICE cannot be measured.

Let's do ICE the right way, there is too much at stake here.

Henry

-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: Wednesday, July 18, 2007 1:19 PM
To: Henry Sinnreich; 'Magnus Westerlund'
Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
Subject: RE: [BEHAVE] Re: ICE deployment data before LC for RFC


> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows=20
> the ICE call flow with two endpoints behind two NAT (Fig. 22=20
> on page 38) require 66 messages just for the SIP INVITE!
>=20
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-sce
> narios-07.txt=20

That figure does not show 66 messages.

That figure is clearly showing how the STUN messages are NATted,
which leads to an incorrect conclusion around the number of=20
messages.  Specifically, for example, it shows this message
exchange which might be counted as "four messages"; yet, there
are only two messages (the allocate request and the allocate
response), which are NATted:

   L               NAT             STUN             =20
                                   Server
   |                |                |              =20
   |(1) Alloc Req   |                |              =20
   |Src=3DL-PRIV-1    |                |              =20
   |Dest=3DSTUN-PUB-1 |                |              =20
   |--------------->|                |              =20
   |                |                |              =20
   |                |(2) Alloc Req   |              =20
   |                |Src=3DL-NAT-PUB-1 |              =20
   |                |Des=3DSTUN-PUB-1  |              =20
   |                |--------------->|              =20
   |                |                |              =20
   |                |(3) Alloc Resp  |              =20
   |                |<---------------|              =20
   |                |Src=3DSTUN-PUB-1  |              =20
   |                |Dest=3DL-NAT-PUB-1|              =20
   |                |Map=3DL-NAT-PUB-1 |              =20
   |                |Rel=3DSTUN-PUB-2  |              =20
   |                |                |              =20
   |(4) Alloc Resp  |                |              =20
   |<---------------|                |              =20
   |Src=3DSTUN-PUB-1  |                |              =20
   |Dest=3DL-PRIV-1   |                |              =20
   |Map=3DL-NAT-PUB-1 |                |              =20
   |Rel=3DSTUN-PUB-2  |                |              =20
   |                |                |              =20

If we added another NAT on that path, the number of messages
on the network does not double.  The number of messages on=20
the network stays the same -- the messages merely have their
IP and UDP headers rewritten by the NATs.  That NAT function
does not increase the number of messages on the network.



Similarly, another message shown in that same figure is also
one message; however, it's shown as two messages:

L               NAT             STUN              NAT               R
                                Server
|(9) SIP INVITE  |                |                |                |
|------------------------------------------------->|                |
|                |                |                |(10) SIP INVITE |
|                |                |                |--------------->|

As you know, there aren't two messages -- it's one message that is
NATted.

If message 9 and 10 were shown as clearly as messages 1-4, messages
9-10 would be shown as four messages (one from L to its NAT, another=20
from NAT to NAT, and the forth (shown) from NAT to R).  But this is
just one SIP message, which is NATted.  It is inaccurate  to count
it as two separate messages or as four separate messages, because
it is only one packet that is NATted.  Counting a STUN packet,
or a DNS packet, as a separate message each time it traverses a=20
NAT is similarly inaccurate.

By my count, figure 22 exchanges 24 STUN/ICE messages and 6=20
SIP messages.

> What happens if the two endpoints are in different ISPs that=20
> have NAT of their own in addition to the ones in Fig. 22?

The same 24 STUN/ICE messages would be exchanged, and the same
6 SIP messages would be exchanged.

> What happens if one would try ICE along a P2PSIP route with=20
> say 5-8 hops?

The SIP signaling messages would traverse those 5-8 additional=20
hops between the P2PSIP nodes participating in the SIP proxy
overlay.  The STUN/ICE messages are sent peer-to-peer, and=20
would remain the same 24 messages if you're doing the same=20
Figure 22 flow. =20

> Would there be hundreds of messages?

No.

-d


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Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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Phil,

> In my view, ICE _is_ a "Hole Punching Technique".

Yes, you absolutely right, though the devil is in the details.

We cannot go here into a shoot-out between hole punching or other
mentioned on the list vs. ICE, but there seems to be a huge difference
in effectiveness and performance.=20

Of course I cannot prove it since there are no measurements published
for ICE, only faith.=20

Just think of traversing all the NAT while going over 5-10 overlay
network hops. Also many ISP are NATed and some are multiple NATed. We
are supposed to provide a solution that works across the Internet. A
provable (testing tools) and proven solution (measurements).

I like ICE and hope to be proven wrong, but the risk is just too big for
now.

Henry

-----Original Message-----
From: Philip Matthews [mailto:philip_matthews@magma.ca]=20
Sent: Wednesday, July 18, 2007 9:27 AM
To: Pyda Srisuresh
Cc: Frank W. Miller; Henry Sinnreich; 'Magnus Westerlund'; sip@ietf.org;
behave@ietf.org; mmusic@ietf.org
Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for
RFC

Frank, Henry, Suresh:

I agree that ICE is a complex spec. I also agree that it would be =20
very interesting to see
more data on which situations result in a direct path between =20
endpoints, as opposed to
a path via a TURN server.

However, I don't understand the comment that "Hole Punching =20
Techniques" are an alternative.
In my view, ICE _is_ a "Hole Punching Technique".

Perhaps you are referring to the use of STUN alone???
The problem with using STUN alone is that there are cases where using =20
the reflexive address
doesn't work, and there is no way with STUN alone to determine when =20
an endpoint should or
should not use a reflexive address.

But perhaps you are referring to something else?
- Philip


On 18-Jul-07, at 11:37 , Pyda Srisuresh wrote:

> Thanks to Henry and Frank for so eloquently articulating this.
>
> As both Henry and Frank mentioned, I fully support standardizing on =20
> HOLE
> PUNCHING TECHNIQUE and leveraging the IETF's considerable influence =20
> to force
> NAT vendors to migrate towards implementations that are friendly to =20
> Hole
> Punching and the Application developers to design applications =20
> using the hole
> punching techniques. That should be the FIRST order of priority.
>
> Yet, what is happening in BEHAVE WG is the other way around. Some =20
> people have
> put their weight so heavily in favor of ICE that they asked to drop =20
> the work on
> hole punching technique as WG item. THIS IS WRONG. I no longer see =20
> "NAT
> Traveral application design guidelines" as BEHAVE WG milestone.
>
> Magnus - Can you explain why the IETF is not allowing HOLE PUNCHING =20
> TECHNIQUES
> to move forward as a WG item?
>
> regards,
> suresh
>
> --- "Frank W. Miller" <fwmiller@cornfed.com> wrote:
>
>>
>>
>> Greetings,
>>
>>
>>
>> What I think Henry is advocating is a systematic approach to =20
>> making sure
>> that the content of the ICE spec is right for its intended =20
>> purpose, i.e. a
>> complete, general NAT traversal solution.  NAT traversal has been =20
>> in my
>> opinion one of the most difficult aspects of SIP implementation =20
>> and its very
>> important that its eventual standardized solution be right.  I =20
>> know I spent
>> a lot of time implementing all the "tests" in the first 3489, only to
>> discover that they were not that useful.  There are two valid =20
>> technical
>> concerns that are put forth in this email:
>>
>>
>>
>> 1)     Viability of correct operation in practical =20
>> implementations.  ICE can
>> result in very complicated message transfers.
>>
>> 2)     Performance.  As pointed out in this email and in other =20
>> places, the
>> additional latency associated with ICE operations can be significant.
>>
>>
>>
>> Henry also points out that there are competing approaches (e.g. Hole
>> Punching) that can perform well in many situations and have better
>> performance.  IMHO, another approach by the IETF in general would =20
>> be to
>> standardize on Hole Punching and leverage the IETF's considerable =20
>> influence
>> to force NAT vendors to migrate towards implementations that are =20
>> friendly to
>> Hole Punching.
>>
>>
>>
>> The bottom line is that I agree with Henry in that labeling ICE as an
>> experimental draft is the right thing to do.  It will allow those
>> implementing ICE (including myself in all likelihood) to proceed =20
>> but signals
>> to the users the correct status of the standard's contents.
>>
>>
>>
>> I also want to applaud loudly the underlying tone of Henry's =20
>> thread here.
>> That is, in solving specific technical problems within the IETF =20
>> forum,
>> focusing on empirical observation of working implementations =20
>> should be the
>> MO that we all strive for when developing and proposing what may =20
>> become
>> protocol standards.
>>
>>
>>
>> Thanks,
>>
>> FM
>>
>>
>>
>>
>>
>>   _____
>>
>> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
>> Sent: Wednesday, July 18, 2007 8:20 AM
>> To: Magnus Westerlund
>> Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
>> Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>>
>>
>> Magnus,
>>
>>
>>
>> I sincerely appreciate your interest and this note!
>>
>>
>>
>>> I don't quite understand why you are requesting a status change =20
>>> of ICE
>>
>>
>>
>> However the arguments stated by the WG co-chairs and by you here are
>> procedural in nature, whereas the facts are some cause for serious =20
>> concern,
>> and I will just mention a few facts:
>>
>>
>>
>> RFC 3489 (STUN) was also presented as a NAT traversal solution =20
>> back in 2003,
>> but <draft-ietf-behave-rfc3489bis-07> now clearly admits that =20
>> "STUN is not a
>> NAT traversal solution by itself. Rather, it is a tool to be used =20
>> in the
>> context of a NAT traversal solution. This is an important change =20
>> from the
>> previous version of this specification (RFC 3489), which presented =20
>> STUN as a
>> complete solution."
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-behave-=20
>> rfc3489bis-07.txt
>>
>> My guess is, in plain English, RFC 3489 turned out just NOT to be an
>> acceptable solution as it was hoped to be.
>>
>>
>>
>> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the =20
>> ICE call
>> flow with two endpoints behind two NAT (Fig. 22 on page 38) =20
>> require 66
>> messages just for the SIP INVITE!
>>
>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-=20
>> scenarios-07.txt
>>
>> What happens if the two endpoints are in different ISPs that have =20
>> NAT of
>> their own in addition to the ones in Fig. 22?
>>
>> What happens if one would try ICE along a P2PSIP route with say =20
>> 5-8 hops?
>> Would there be hundreds of messages? What is the likelihood for =20
>> failure?
>>
>>
>>
>> By contrast with such approaches based on hope and faith, the =20
>> BEHAVE WG has
>> produced work based on real measurements and tools to do the =20
>> measurements,
>> such as
>>
>>
>>
>> "NAT Classification Test Results"
>>
>>
>>
>> "Application Design Guidelines for Traversal through Network Address
>> Translators"
>>
>> The measurements for hole punching are reported in
>>
>> "Peer-to-Peer Communication Across Network Address Translators"
>>
>> http://www.brynosaurus.com/pub/net/p2pnat/
>>
>>
>>
>> I am also a believer in ICE and have actually urged Jonathan =20
>> Rosenberg to do
>> this work and he has given his best. Now that we have ICE-17 =20
>> version, it is
>> in the best interest of the industry and the IETF to make it an =20
>> experimental
>> RFC, but not yet a standard so that:
>>
>>  1. Independent, compliant implementations will be developed,
>>
>>  2. Testing in events such as SIPit,
>>
>>  3. Reporting the results, such as reported for the "hole =20
>> punching" above.
>>
>>
>>
>> What happens if indeed hole punching proves to be the far more =20
>> effective
>> approach?
>>
>>
>>
>> The industry can ill afford a new series of RFC[ICE]-bis and all =20
>> the years
>> it takes to move from believing to measuring and proving. I am =20
>> sure IETF
>> procedures were meant in this spirit (though I am not good in =20
>> producing the
>> right quotes).
>>
>>
>>
>> Thanks again for your attention to this concern about ICE.
>>
>> Also my apology to all the anguish to the numerous contributors to =20
>> the ICE
>> related I-Ds, but writing test tools and reporting measurements is =20
>> the right
>> way to go.
>>
>>
>>
>> Henry
>>
>>
>>
>>
>>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Wednesday, July 18, 2007 1:11 AM
>> To: Henry Sinnreich
>> Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
>> Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>>
>>
>> Henry Sinnreich skrev:
>>
>>> Following some discussions and soul searching, the best approach for
>>
>>> ICE-17 for LC would be to _go ahead and make it an experimental =20
>>> RFC_.
>>
>>>
>>
>>> This would support the development of interoperable =20
>>> implementations, the
>>
>>> collection of deployment data and also hopefully open source code.
>>
>>>
>>
>>> Though I have full faith in ICE, following the practices that =20
>>> made the
>>
>>> IETF successful take precedent here, I believe.
>>
>>>
>>
>>> This is a change from the attached that I wrote on 7/16/2007.
>>
>>>
>>
>>> Note: We can only hope the proliferation of SBC in VoIP service =20
>>> provider
>>
>>> networks will not make these concerns and ICE for SIP a moot =20
>>> issue, but
>>
>>> this is another topic.
>>
>>
>>
>> Henry,
>>
>>
>>
>> I don't quite understand why you are requesting a status change of =20
>> ICE?
>>
>> To me it appears that ICE is almost finished spec. It has been
>>
>> implemented, used and feedback has been influencing the protocol. =20
>> It is
>>
>> definitly one of the better examples of current IETF work that =20
>> actually
>>
>> has running code. Sure, the implementations may not be fully =20
>> updated yet
>>
>> to the latest draft version. But I think it is safe to say that ICE
>>
>> works in the intended core cases. If there are some corner cases =20
>> where
>>
>> it fails, that may be. But that is not information we will have until
>>
>> really large scale deployement or someone makes very extensive tests.
>>
>>
>>
>> I think the IETF and the Internet has much more benefit from a =20
>> standards
>>
>> track solution than any of the risks existing with ICE of today.
>>
>>
>>
>> And to be clear, proposed standard is allowed to be published =20
>> without a
>>
>> single implementation exists. It might not be good practice, but =20
>> it is
>>
>> allowed by our process. And as I said before ICE has had enough
>>
>> implementation that I am very confortable in balloting YES for it =20
>> when
>>
>> it appears on the IESGs table.
>>
>>
>>
>> Cheers
>>
>>
>>
>> Magnus Westerlund
>>
>>
>>
>> IETF Transport Area Director & TSVWG Chair
>>
>> ---------------------------------------------------------------------

>> -
>>
>> Multimedia Technologies, Ericsson Research EAB/TVM/M
>>
>> ---------------------------------------------------------------------

>> -
>>
>> Ericsson AB                | Phone +46 8 4048287
>>
>> Torshamsgatan 23           | Fax   +46 8 7575550
>>
>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>> ---------------------------------------------------------------------

>> -
>>
>>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
>>
>
>
>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave



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Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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=20
> * Publish open source code at Sourceforge.net or similar,
> * Publish a test tool as well,
> * Gather and publish the results, possibly involving SIPit or another,
> * Perfect the OS code and the protocol until it works OK.
>=20
> This has been done for "Hole Punching for P2P" in=20
> http://www.brynosaurus.com/pub/net/p2pnat/=20

Heuh, no. This is just "exploratory concept" stuff, certainly
not an interoperable protocol specification.

And if the proponents of this alternative hole punching didn't=20
propose a draft at IETF when we starting working on this back
in 2001, it seems to me that it's too late.

> My assumption is that hole punching is much more effective=20
> for P2PSIP and I have copied the P2PSIP WG on this. The ratio=20
> of messages is probably 5:1 or higher for ICE vs. hole=20
> punching and this is critical when routing a P2PSIP message=20
> over an overlay in 5-10 hops. Of course I cannot prove it=20
> since as yet ICE cannot be measured.

Well, your assumption may be wrong, especially in the absence of
a specification for this alternative hole punching technique.

The other thing I find funny in here is that the alternative=20
hole punching techique as described on the web site uses a
centralize server common to BOTH ends to relay the address, while
ICE does not (i.e., each end can use their own STUN server).

In other words, the major difference between ICE and the
alternative-hole-
punching technique seems to be that alternative-hole-punching
assumes both ends use the same STUN server, which makes the=20
rest of the exchange simpler (the "magic box" architectural
concept is always handy).

Ironically, this seems to me to be LESS usable in a global network
where you wont't use centralize servers, especially for p2psip.



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Fine with me.

Janet

"Audet, Francois (SC100:3055)" <audet@nortel.com> wrote on 07/18/2007 
02:58:39 PM:

> I will be happy if the document is revised to explain that you 
> cannot expect that responses received by a UAC or Proxy will 
> necessarily include RPM, in particular in systems where forking 
> occurs, and that therefore, applications that use this draft must 
> take it into account.
> 
> From: Janet P Gunn [mailto:jgunn6@csc.com] 
> Sent: Wednesday, July 18, 2007 10:41
> To: Audet, Francois (SC100:3055)
> Cc: sip@ietf.org; James M. Polk
> Subject: RE: [Sip] New draft modifying RFC 4412 for Responses

> 
> 
> In line. 
> 
> "Francois Audet" <audet@nortel.com> wrote on 07/18/2007 11:26:54 AM:
> 
> > > > What I mean is that the error response may not be delivered 
> > > to the UAC 
> > > > because a forking proxy will only deliver one final 
> > > response. If there 
> > > > are multiple error responses on the individual forks, one being 
the 
> > > > one with the RPH, and the other being some other response, 
> > > it is up to 
> > > > the proxy to decide which one to forward back.
> > > > 
> > > > So the UAC can not rely on receiving the RPH to "re-intiate" the 
> > > > session or provide some alternative treatment. This may or 
> > > may not be 
> > > > a problem depending on the application (that's why I said "if it 
> > > > absolutely has").
> > > > 
> > > 
> > > I don't think that will be a problem. 
> > > 
> > > For one thing (for the namespaces I am working with) the 
> > > expectation would be that ALL the responses (except 100 and 
> > > 403) would have the RPH on the response, so it wouldn't 
> > > matter which one made it back to the UAC. 
> > 
> > I am not talking about multiple provisional responses from the same
> > endpoint.
> > 
> > I am talking about multiple responses (provisional and final) from 
> > multiple forks. For example, multiple 4XX responses. 
> 
> Yes, I understood that.
> > 
> > While it is possible that a single forking proxy may decide to 
> > include the RPH in all responses, that proxy has absolutely no 
> > control over downstream and upstream proxies who may not include it
> > (they may be proxies in different domains).
> > 
> > That's the point I'm trying to make. You can not enforce the 
> > presence RPH in responses in a global system. 
> 
> Agree. For the context I am working in, a "global system" is not of 
> concern.  I would not expect universal "RPH in responses" in a global 
system.
> 
> > 
> > > Secondly, the expectation is that the devices responsible for 
> > > "re-initiating" WOULD be stateful, and not dependant on the 
> > > information in the RPH in the response.  It is the 
> > > intervening devices that might not be stateful, and would 
> > > take advantage of the RPH information in the responses. 
> > 
> > That is precisely the type of clarification that needs to be included
> > in the draft. It points out the constraints of the mechanism so 
> > implementors use the feature properly.
> > 
> > Some text explaining that the UAC needs to be able to deal with the 
RPH
> > not being present in responses gracefully (and why) is needed. 
> 
> I am not sure why that needs to be in the ID.  It is an 
> implementation specific issue. 
> 
> Since  RFC 4412, as currently published, forbids RPH in responses, I
> think it "goes without saying" that "the UAC needs to be able to 
> deal with the RPH not being present in responses". 
> 
> Further, while I know that the systems (and namespaces) _I_ work 
> with intend to use RPH in responses, I also have been told that 
> other systems (using other namespaces) do not intend to put RPH in 
> ANY responses.  I do not think there is any need to standardize this
> aspect, just to adjust the standard so it is PERMITTED.
> > 
> > Similar text for proxies also need to be there. What if a proxy 
> > forwards a request with RPH and it receives nultiple responses from
> > the downstream proxies, with some including the RPH and others not
> > including it? Does this need to be standardized? 
> 
> Again, I do not think there is a need to standardize this aspect. 
> 
> Janet
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
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<br><font size=2 face="sans-serif"><br>
<br>
Fine with me.</font>
<br>
<br><font size=2 face="sans-serif">Janet</font>
<br>
<br><font size=2><tt>&quot;Audet, Francois (SC100:3055)&quot; &lt;audet@nortel.com&gt;
wrote on 07/18/2007 02:58:39 PM:<br>
<br>
&gt; I will be happy if the document is revised to explain that you <br>
&gt; cannot expect that responses received by a UAC or Proxy will <br>
&gt; necessarily include RPM, in particular in systems where forking <br>
&gt; occurs, and that therefore, applications that use this draft must
<br>
&gt; take it into account.</tt></font>
<br><font size=2><tt>&gt; <br>
&gt; From: Janet P Gunn [mailto:jgunn6@csc.com] <br>
&gt; Sent: Wednesday, July 18, 2007 10:41<br>
&gt; To: Audet, Francois (SC100:3055)<br>
&gt; Cc: sip@ietf.org; James M. Polk<br>
&gt; Subject: RE: [Sip] New draft modifying RFC 4412 for Responses<br>
</tt></font>
<br><font size=2><tt>&gt; <br>
&gt; <br>
&gt; In line. <br>
&gt; <br>
&gt; &quot;Francois Audet&quot; &lt;audet@nortel.com&gt; wrote on 07/18/2007
11:26:54 AM:<br>
&gt; <br>
&gt; &gt; &gt; &gt; What I mean is that the error response may not be delivered
<br>
&gt; &gt; &gt; to the UAC <br>
&gt; &gt; &gt; &gt; because a forking proxy will only deliver one final
<br>
&gt; &gt; &gt; response. If there <br>
&gt; &gt; &gt; &gt; are multiple error responses on the individual forks,
one being the <br>
&gt; &gt; &gt; &gt; one with the RPH, and the other being some other response,
<br>
&gt; &gt; &gt; it is up to <br>
&gt; &gt; &gt; &gt; the proxy to decide which one to forward back.<br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; &gt; So the UAC can not rely on receiving the RPH to &quot;re-intiate&quot;
the <br>
&gt; &gt; &gt; &gt; session or provide some alternative treatment. This
may or <br>
&gt; &gt; &gt; may not be <br>
&gt; &gt; &gt; &gt; a problem depending on the application (that's why
I said &quot;if it <br>
&gt; &gt; &gt; &gt; absolutely has&quot;).<br>
&gt; &gt; &gt; &gt; <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; I don't think that will be a problem. <br>
&gt; &gt; &gt; <br>
&gt; &gt; &gt; For one thing (for the namespaces I am working with) the
<br>
&gt; &gt; &gt; expectation would be that ALL the responses (except 100
and <br>
&gt; &gt; &gt; 403) would have the RPH on the response, so it wouldn't
<br>
&gt; &gt; &gt; matter which one made it back to the UAC. <br>
&gt; &gt; <br>
&gt; &gt; I am not talking about multiple provisional responses from the
same<br>
&gt; &gt; endpoint.<br>
&gt; &gt; <br>
&gt; &gt; I am talking about multiple responses (provisional and final)
from <br>
&gt; &gt; multiple forks. For example, multiple 4XX responses. <br>
&gt; <br>
&gt; Yes, I understood that.<br>
&gt; &gt; <br>
&gt; &gt; While it is possible that a single forking proxy may decide to
<br>
&gt; &gt; include the RPH in all responses, that proxy has absolutely no
<br>
&gt; &gt; control over downstream and upstream proxies who may not include
it<br>
&gt; &gt; (they may be proxies in different domains).<br>
&gt; &gt; <br>
&gt; &gt; That's the point I'm trying to make. You can not enforce the
<br>
&gt; &gt; presence RPH in responses in a global system. <br>
&gt; <br>
&gt; Agree. For the context I am working in, a &quot;global system&quot;
is not of <br>
&gt; concern. &nbsp;I would not expect universal &quot;RPH in responses&quot;
in a global system.<br>
&gt; <br>
&gt; &gt; <br>
&gt; &gt; &gt; Secondly, the expectation is that the devices responsible
for <br>
&gt; &gt; &gt; &quot;re-initiating&quot; WOULD be stateful, and not dependant
on the <br>
&gt; &gt; &gt; information in the RPH in the response. &nbsp;It is the
<br>
&gt; &gt; &gt; intervening devices that might not be stateful, and would
<br>
&gt; &gt; &gt; take advantage of the RPH information in the responses.
<br>
&gt; &gt; <br>
&gt; &gt; That is precisely the type of clarification that needs to be
included<br>
&gt; &gt; in the draft. It points out the constraints of the mechanism
so <br>
&gt; &gt; implementors use the feature properly.<br>
&gt; &gt; <br>
&gt; &gt; Some text explaining that the UAC needs to be able to deal with
the RPH<br>
&gt; &gt; not being present in responses gracefully (and why) is needed.
<br>
&gt; <br>
&gt; I am not sure why that needs to be in the ID. &nbsp;It is an <br>
&gt; implementation specific issue. &nbsp; <br>
&gt; <br>
&gt; Since &nbsp;RFC 4412, as currently published, forbids RPH in responses,
I<br>
&gt; think it &quot;goes without saying&quot; that &quot;the UAC needs
to be able to <br>
&gt; deal with the RPH not being present in responses&quot;. <br>
&gt; <br>
&gt; Further, while I know that the systems (and namespaces) _I_ work <br>
&gt; with intend to use RPH in responses, I also have been told that <br>
&gt; other systems (using other namespaces) do not intend to put RPH in
<br>
&gt; ANY responses. &nbsp;I do not think there is any need to standardize
this<br>
&gt; aspect, just to adjust the standard so it is PERMITTED.<br>
&gt; &gt; <br>
&gt; &gt; Similar text for proxies also need to be there. What if a proxy
<br>
&gt; &gt; forwards a request with RPH and it receives nultiple responses
from<br>
&gt; &gt; the downstream proxies, with some including the RPH and others
not<br>
&gt; &gt; including it? Does this need to be standardized? <br>
&gt; <br>
&gt; Again, I do not think there is a need to standardize this aspect.
<br>
&gt; <br>
&gt; Janet<br>
&gt; &gt; <br>
&gt; &gt; <br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; Sip mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip<br>
&gt; &gt; This list is for NEW development of the core SIP Protocol<br>
&gt; &gt; Use sip-implementors@cs.columbia.edu for questions on current
sip<br>
&gt; &gt; Use sipping@ietf.org for new developments on the application
of sip</tt></font>
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Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
Date: Wed, 18 Jul 2007 12:42:57 -0700
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
To: "Francois Audet" <audet@nortel.com>, "Dan Wing" <dwing@cisco.com>,
	"Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Francois,

> Heuh, no. This is just "exploratory concept" stuff, certainly not an=20
> interoperable protocol specification.

I am happy to report we are in total agreement.

Henry

-----Original Message-----
From: Francois Audet [mailto:audet@nortel.com]=20
Sent: Wednesday, July 18, 2007 2:38 PM
To: Henry Sinnreich; Dan Wing; Magnus Westerlund
Cc: sip@ietf.org; p2psip@ietf.org; behave@ietf.org; Philip Matthews;
mmusic@ietf.org
Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC

=20
> * Publish open source code at Sourceforge.net or similar,
> * Publish a test tool as well,
> * Gather and publish the results, possibly involving SIPit or another,
> * Perfect the OS code and the protocol until it works OK.
>=20
> This has been done for "Hole Punching for P2P" in=20
> http://www.brynosaurus.com/pub/net/p2pnat/=20

Heuh, no. This is just "exploratory concept" stuff, certainly
not an interoperable protocol specification.

And if the proponents of this alternative hole punching didn't=20
propose a draft at IETF when we starting working on this back
in 2001, it seems to me that it's too late.

> My assumption is that hole punching is much more effective=20
> for P2PSIP and I have copied the P2PSIP WG on this. The ratio=20
> of messages is probably 5:1 or higher for ICE vs. hole=20
> punching and this is critical when routing a P2PSIP message=20
> over an overlay in 5-10 hops. Of course I cannot prove it=20
> since as yet ICE cannot be measured.

Well, your assumption may be wrong, especially in the absence of
a specification for this alternative hole punching technique.

The other thing I find funny in here is that the alternative=20
hole punching techique as described on the web site uses a
centralize server common to BOTH ends to relay the address, while
ICE does not (i.e., each end can use their own STUN server).

In other words, the major difference between ICE and the
alternative-hole-
punching technique seems to be that alternative-hole-punching
assumes both ends use the same STUN server, which makes the=20
rest of the exchange simpler (the "magic box" architectural
concept is always handy).

Ironically, this seems to me to be LESS usable in a global network
where you wont't use centralize servers, especially for p2psip.



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From sip-bounces@ietf.org Wed Jul 18 16:26:36 2007
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Subject: Re: [Sip] WGLC for draft-ietf-sip-sips-05.txt
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A little late, but one consideration:

Deprecating the last hop exception means that if the last hop proxy supports 
this specification, the UAC has no way to request SIPS up to the last hop 
(which it may want to do; best-effort TLS using sip: will probably not work 
in the intermediate period until this specification is widely supported).

Would it be an idea for the draft to define request disposition tags 
(RFC3841) "last-hop-exception" and "no-last-hop-exception"? Then the UA 
could use SIPS, but indicate its preference to allow the last hop exception, 
and get "almost-end-2-end" security

Regards,
Jeroen

Dean Willis wrote:
> I'm pleased to announce that the authors think that the SIPS draft is
> ready for working group last call. This draft has had the privilege
> of more review prior to WGLC than most of our other efforts, so
> perhaps it won't be too controversial.
>
> Please make a final read through the draft and bring any comments
> back to this list.
>
> I'd like to conclude this WGLC by July 16, 2007, so get your comments
> back in by then. And I'll plan to put some discussion time on the
> agenda in Chicago.
>
> Thanks, and have a great weekend. 



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From sip-bounces@ietf.org Wed Jul 18 16:57:11 2007
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Subject: RE: [Sip] WGLC for draft-ietf-sip-sips-05.txt
Date: Wed, 18 Jul 2007 15:56:44 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>, "IETF SIP" <sip@ietf.org>,
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We have explored something similar (using an option tag instead) in an
earlier version (see=20
https://svn.resiprocate.org/rep/ietf-drafts/audet/draft-ietf-sip-sips-03
.txt),=20
but people felt the additional complexity was not warranted and that
allowing for the last-hop exception was not a good goal.  It was
therefore revoved in the next version.

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
> Sent: Wednesday, July 18, 2007 13:23
> To: IETF SIP; Dean Willis; Audet, Francois (SC100:3055)
> Subject: Re: [Sip] WGLC for draft-ietf-sip-sips-05.txt
>=20
> A little late, but one consideration:
>=20
> Deprecating the last hop exception means that if the last hop=20
> proxy supports this specification, the UAC has no way to=20
> request SIPS up to the last hop (which it may want to do;=20
> best-effort TLS using sip: will probably not work in the=20
> intermediate period until this specification is widely supported).
>=20
> Would it be an idea for the draft to define request disposition tags
> (RFC3841) "last-hop-exception" and "no-last-hop-exception"?=20
> Then the UA could use SIPS, but indicate its preference to=20
> allow the last hop exception, and get "almost-end-2-end" security
>=20
> Regards,
> Jeroen
>=20
> Dean Willis wrote:
> > I'm pleased to announce that the authors think that the=20
> SIPS draft is=20
> > ready for working group last call. This draft has had the=20
> privilege of=20
> > more review prior to WGLC than most of our other efforts,=20
> so perhaps=20
> > it won't be too controversial.
> >
> > Please make a final read through the draft and bring any=20
> comments back=20
> > to this list.
> >
> > I'd like to conclude this WGLC by July 16, 2007, so get=20
> your comments=20
> > back in by then. And I'll plan to put some discussion time on the=20
> > agenda in Chicago.
> >
> > Thanks, and have a great weekend.=20
>=20
>=20


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From sip-bounces@ietf.org Wed Jul 18 17:06:02 2007
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Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
From: "Frank W. Miller" <fwmiller@cornfed.com>
To: Francois Audet <audet@nortel.com>
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On Wed, 2007-07-18 at 14:38 -0500, Francois Audet wrote:

> The other thing I find funny in here is that the alternative 
> hole punching techique as described on the web site uses a
> centralize server common to BOTH ends to relay the address, while
> ICE does not (i.e., each end can use their own STUN server).
> 
> In other words, the major difference between ICE and the
> alternative-hole-
> punching technique seems to be that alternative-hole-punching
> assumes both ends use the same STUN server, which makes the 
> rest of the exchange simpler (the "magic box" architectural
> concept is always handy).
> 
> Ironically, this seems to me to be LESS usable in a global network
> where you wont't use centralize servers, especially for p2psip.
> 

This may be sufficient, but I don't think its necessary.  Each UA
punches the hole (and keeps it open, as David described) to whatever
STUN server they like, all that's required is that each server have the
same view of each client.  That is somewhat less restrictive.  In the
common case of the open Internet, as long as the STUN servers used by
each respective UA are in the "open", they don't have to be the same.
After that, the "public" or "visible" IP address/port is put in the
Contact headers.

Regarding the 90% coverage, this is what I'm referring to when I say
that it may be preferable to influence the NAT vendors to not do the
things that put them in the 10% cases.  I sort of thought that was what
the NAT guidelines draft was for.

In addition, if we are standardizing a protocol based on the 10% cases
and imposing potentially huge costs on the 90% to do so, then our
approach is incorrect.  We should be designing for the 90% case and
handling the 10% as exceptional behavior, i.e. a solution where you pay
the big cost only in the case where you are actually in the 10%.

FM




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From sip-bounces@ietf.org Wed Jul 18 18:39:17 2007
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Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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> -----Original Message-----
> From: Melinda Shore [mailto:mshore@cisco.com]
> 
> ICE makes me want to pound red-hot nails into my eye sockets.  That said,
> 90% really isn't nearly good enough for residential deployments.

Incidentally, 90% is where my stats come out today -- this doesn't include
restricted-cone/symmetric and symmetric/symmetric penetration, which has
been prototyped but not added to the production system yet.  The hope is
this will get us much close to the 99% range, but it remains to be seen by
how much.

Furthermore, I'd wager that for most applications (such as VoIP), even
routing 1 in 10 calls through a central relay isn't going to break the bank.
(Or for mega-discounts, use the full-cone and non-NAT'd clients (>30%) to
act as "free" relays for their peers.)

Thus while NAT penetration using what I guess we're calling the "hole
punching technique" has been demonstrated to work at 90%+ on a very large
scale (hundreds of thousands of clients tested) we should expect that it's
possible to get even better still.  If ICE can show how it'd do better,
that'd be great.  But we can't tolerate ICE being any worse than that, and
thus before rushing ahead with standardizing an unproven ICE, we should just
take the time to prove it.

-david



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From sip-bounces@ietf.org Wed Jul 18 18:54:43 2007
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Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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I agree with Henry, Frank, Suresh, and the lurkers on this list who have
concern that ICE hasn't demonstrated its claims in the real world, while
others have achieved similar ends with great success using much simpler
approaches.  (Some examples: Skype, Hamachi, Red Swoosh/Akamai, etc.)

For example, regarding the "hole punching" approach (by which I assume you
mean "opening a hole and trying to keep it open" versus ICE's "open a hole
'on demand' for each connection"), it's been demonstrated to work at large
scale with at least 90% direct peer connectivity across the internet, today,
with a much simpler message flow and implementation.  I find it concerning
that ICE has taken a more costly, difficult route without confirming it's
truly worth the additional incremental effort (or even a step backwards).

I share the concern that ICE standardization has gotten far out ahead of the
horse in inventing a new protocol from scratch rather than modeling off of
the proven protocols that came before it, which historically has been the
IETF strength.  Given this choice to go a new route, I think it warrants a
higher burden of evidence -- contrasted against known success rates of
competing solutions -- before being formally endorsed by the IETF.

I'm completely open to ICE being superior to all others.  I have no doubt
more effort has been put into it, and I'm eager to enjoy the fruits of
everyone's labor.  But I'd like to see the data that proves it, and I see no
reason to charge ahead with standardization until data is firmly in hand.

-david

> -----Original Message-----
> From: Philip Matthews [mailto:philip_matthews@magma.ca]
> Sent: Wednesday, July 18, 2007 9:27 AM
> To: Pyda Srisuresh
> Cc: behave@ietf.org; mmusic@ietf.org; sip@ietf.org; 'Henry Sinnreich'
> Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> Frank, Henry, Suresh:
> 
> I agree that ICE is a complex spec. I also agree that it would be
> very interesting to see
> more data on which situations result in a direct path between
> endpoints, as opposed to
> a path via a TURN server.
> 
> However, I don't understand the comment that "Hole Punching
> Techniques" are an alternative.
> In my view, ICE _is_ a "Hole Punching Technique".
> 
> Perhaps you are referring to the use of STUN alone???
> The problem with using STUN alone is that there are cases where using
> the reflexive address
> doesn't work, and there is no way with STUN alone to determine when
> an endpoint should or
> should not use a reflexive address.
> 
> But perhaps you are referring to something else?
> - Philip
> 
> 
> On 18-Jul-07, at 11:37 , Pyda Srisuresh wrote:
> 
> > Thanks to Henry and Frank for so eloquently articulating this.
> >
> > As both Henry and Frank mentioned, I fully support standardizing on
> > HOLE
> > PUNCHING TECHNIQUE and leveraging the IETF's considerable influence
> > to force
> > NAT vendors to migrate towards implementations that are friendly to
> > Hole
> > Punching and the Application developers to design applications
> > using the hole
> > punching techniques. That should be the FIRST order of priority.
> >
> > Yet, what is happening in BEHAVE WG is the other way around. Some
> > people have
> > put their weight so heavily in favor of ICE that they asked to drop
> > the work on
> > hole punching technique as WG item. THIS IS WRONG. I no longer see
> > "NAT
> > Traveral application design guidelines" as BEHAVE WG milestone.
> >
> > Magnus - Can you explain why the IETF is not allowing HOLE PUNCHING
> > TECHNIQUES
> > to move forward as a WG item?
> >
> > regards,
> > suresh
> >
> > --- "Frank W. Miller" <fwmiller@cornfed.com> wrote:
> >
> >>
> >>
> >> Greetings,
> >>
> >>
> >>
> >> What I think Henry is advocating is a systematic approach to
> >> making sure
> >> that the content of the ICE spec is right for its intended
> >> purpose, i.e. a
> >> complete, general NAT traversal solution.  NAT traversal has been
> >> in my
> >> opinion one of the most difficult aspects of SIP implementation
> >> and its very
> >> important that its eventual standardized solution be right.  I
> >> know I spent
> >> a lot of time implementing all the "tests" in the first 3489, only to
> >> discover that they were not that useful.  There are two valid
> >> technical
> >> concerns that are put forth in this email:
> >>
> >>
> >>
> >> 1)     Viability of correct operation in practical
> >> implementations.  ICE can
> >> result in very complicated message transfers.
> >>
> >> 2)     Performance.  As pointed out in this email and in other
> >> places, the
> >> additional latency associated with ICE operations can be significant.
> >>
> >>
> >>
> >> Henry also points out that there are competing approaches (e.g. Hole
> >> Punching) that can perform well in many situations and have better
> >> performance.  IMHO, another approach by the IETF in general would
> >> be to
> >> standardize on Hole Punching and leverage the IETF's considerable
> >> influence
> >> to force NAT vendors to migrate towards implementations that are
> >> friendly to
> >> Hole Punching.
> >>
> >>
> >>
> >> The bottom line is that I agree with Henry in that labeling ICE as an
> >> experimental draft is the right thing to do.  It will allow those
> >> implementing ICE (including myself in all likelihood) to proceed
> >> but signals
> >> to the users the correct status of the standard's contents.
> >>
> >>
> >>
> >> I also want to applaud loudly the underlying tone of Henry's
> >> thread here.
> >> That is, in solving specific technical problems within the IETF
> >> forum,
> >> focusing on empirical observation of working implementations
> >> should be the
> >> MO that we all strive for when developing and proposing what may
> >> become
> >> protocol standards.
> >>
> >>
> >>
> >> Thanks,
> >>
> >> FM
> >>
> >>
> >>
> >>
> >>
> >>   _____
> >>
> >> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> >> Sent: Wednesday, July 18, 2007 8:20 AM
> >> To: Magnus Westerlund
> >> Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
> >> Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> >>
> >>
> >>
> >> Magnus,
> >>
> >>
> >>
> >> I sincerely appreciate your interest and this note!
> >>
> >>
> >>
> >>> I don't quite understand why you are requesting a status change
> >>> of ICE
> >>
> >>
> >>
> >> However the arguments stated by the WG co-chairs and by you here are
> >> procedural in nature, whereas the facts are some cause for serious
> >> concern,
> >> and I will just mention a few facts:
> >>
> >>
> >>
> >> RFC 3489 (STUN) was also presented as a NAT traversal solution
> >> back in 2003,
> >> but <draft-ietf-behave-rfc3489bis-07> now clearly admits that
> >> "STUN is not a
> >> NAT traversal solution by itself. Rather, it is a tool to be used
> >> in the
> >> context of a NAT traversal solution. This is an important change
> >> from the
> >> previous version of this specification (RFC 3489), which presented
> >> STUN as a
> >> complete solution."
> >>
> >> http://www.ietf.org/internet-drafts/draft-ietf-behave-
> >> rfc3489bis-07.txt
> >>
> >> My guess is, in plain English, RFC 3489 turned out just NOT to be an
> >> acceptable solution as it was hoped to be.
> >>
> >>
> >>
> >> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the
> >> ICE call
> >> flow with two endpoints behind two NAT (Fig. 22 on page 38)
> >> require 66
> >> messages just for the SIP INVITE!
> >>
> >> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-
> >> scenarios-07.txt
> >>
> >> What happens if the two endpoints are in different ISPs that have
> >> NAT of
> >> their own in addition to the ones in Fig. 22?
> >>
> >> What happens if one would try ICE along a P2PSIP route with say
> >> 5-8 hops?
> >> Would there be hundreds of messages? What is the likelihood for
> >> failure?
> >>
> >>
> >>
> >> By contrast with such approaches based on hope and faith, the
> >> BEHAVE WG has
> >> produced work based on real measurements and tools to do the
> >> measurements,
> >> such as
> >>
> >>
> >>
> >> "NAT Classification Test Results"
> >>
> >>
> >>
> >> "Application Design Guidelines for Traversal through Network Address
> >> Translators"
> >>
> >> The measurements for hole punching are reported in
> >>
> >> "Peer-to-Peer Communication Across Network Address Translators"
> >>
> >> http://www.brynosaurus.com/pub/net/p2pnat/
> >>
> >>
> >>
> >> I am also a believer in ICE and have actually urged Jonathan
> >> Rosenberg to do
> >> this work and he has given his best. Now that we have ICE-17
> >> version, it is
> >> in the best interest of the industry and the IETF to make it an
> >> experimental
> >> RFC, but not yet a standard so that:
> >>
> >>  1. Independent, compliant implementations will be developed,
> >>
> >>  2. Testing in events such as SIPit,
> >>
> >>  3. Reporting the results, such as reported for the "hole
> >> punching" above.
> >>
> >>
> >>
> >> What happens if indeed hole punching proves to be the far more
> >> effective
> >> approach?
> >>
> >>
> >>
> >> The industry can ill afford a new series of RFC[ICE]-bis and all
> >> the years
> >> it takes to move from believing to measuring and proving. I am
> >> sure IETF
> >> procedures were meant in this spirit (though I am not good in
> >> producing the
> >> right quotes).
> >>
> >>
> >>
> >> Thanks again for your attention to this concern about ICE.
> >>
> >> Also my apology to all the anguish to the numerous contributors to
> >> the ICE
> >> related I-Ds, but writing test tools and reporting measurements is
> >> the right
> >> way to go.
> >>
> >>
> >>
> >> Henry
> >>
> >>
> >>
> >>
> >>
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Wednesday, July 18, 2007 1:11 AM
> >> To: Henry Sinnreich
> >> Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
> >> Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> >>
> >>
> >>
> >> Henry Sinnreich skrev:
> >>
> >>> Following some discussions and soul searching, the best approach for
> >>
> >>> ICE-17 for LC would be to _go ahead and make it an experimental
> >>> RFC_.
> >>
> >>>
> >>
> >>> This would support the development of interoperable
> >>> implementations, the
> >>
> >>> collection of deployment data and also hopefully open source code.
> >>
> >>>
> >>
> >>> Though I have full faith in ICE, following the practices that
> >>> made the
> >>
> >>> IETF successful take precedent here, I believe.
> >>
> >>>
> >>
> >>> This is a change from the attached that I wrote on 7/16/2007.
> >>
> >>>
> >>
> >>> Note: We can only hope the proliferation of SBC in VoIP service
> >>> provider
> >>
> >>> networks will not make these concerns and ICE for SIP a moot
> >>> issue, but
> >>
> >>> this is another topic.
> >>
> >>
> >>
> >> Henry,
> >>
> >>
> >>
> >> I don't quite understand why you are requesting a status change of
> >> ICE?
> >>
> >> To me it appears that ICE is almost finished spec. It has been
> >>
> >> implemented, used and feedback has been influencing the protocol.
> >> It is
> >>
> >> definitly one of the better examples of current IETF work that
> >> actually
> >>
> >> has running code. Sure, the implementations may not be fully
> >> updated yet
> >>
> >> to the latest draft version. But I think it is safe to say that ICE
> >>
> >> works in the intended core cases. If there are some corner cases
> >> where
> >>
> >> it fails, that may be. But that is not information we will have until
> >>
> >> really large scale deployement or someone makes very extensive tests.
> >>
> >>
> >>
> >> I think the IETF and the Internet has much more benefit from a
> >> standards
> >>
> >> track solution than any of the risks existing with ICE of today.
> >>
> >>
> >>
> >> And to be clear, proposed standard is allowed to be published
> >> without a
> >>
> >> single implementation exists. It might not be good practice, but
> >> it is
> >>
> >> allowed by our process. And as I said before ICE has had enough
> >>
> >> implementation that I am very confortable in balloting YES for it
> >> when
> >>
> >> it appears on the IESGs table.
> >>
> >>
> >>
> >> Cheers
> >>
> >>
> >>
> >> Magnus Westerlund
> >>
> >>
> >>
> >> IETF Transport Area Director & TSVWG Chair
> >>
> >> ---------------------------------------------------------------------
> >> -
> >>
> >> Multimedia Technologies, Ericsson Research EAB/TVM/M
> >>
> >> ---------------------------------------------------------------------
> >> -
> >>
> >> Ericsson AB                | Phone +46 8 4048287
> >>
> >> Torshamsgatan 23           | Fax   +46 8 7575550
> >>
> >> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >>
> >> ---------------------------------------------------------------------
> >> -
> >>
> >>> _______________________________________________
> >> Behave mailing list
> >> Behave@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/behave
> >>
> >
> >
> >
> >
> >
> > _______________________________________________
> > Behave mailing list
> > Behave@ietf.org
> > https://www1.ietf.org/mailman/listinfo/behave
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jul 18 18:54:45 2007
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Date: Wed, 18 Jul 2007 11:38:24 -0700 (PDT)
From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
To: Philip Matthews <philip_matthews@magma.ca>
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--- Philip Matthews <philip_matthews@magma.ca> wrote:

> Frank, Henry, Suresh:
> 
> I agree that ICE is a complex spec. I also agree that it would be  
> very interesting to see
> more data on which situations result in a direct path between  
> endpoints, as opposed to
> a path via a TURN server.
> 
> However, I don't understand the comment that "Hole Punching  
> Techniques" are an alternative.
> In my view, ICE _is_ a "Hole Punching Technique".
> 
> Perhaps you are referring to the use of STUN alone???
> The problem with using STUN alone is that there are cases where using  
> the reflexive address
> doesn't work, and there is no way with STUN alone to determine when  
> an endpoint should or
> should not use a reflexive address.
> 

[suresh] Right, ICE does use the HOLE PUNCHING technique and the BCP guidelines
described in draft-ford-behave-app draft, even though ICE draft does NOT
acknowledge or reference the above draft or prior work that introduced HOLE
PUNCHING TECHNIQUE. ICE is also using HOLE PUNCHING in a limited way within the
constraints of STUN and SIP. Hence, the croner case scenarios you site. ICE is
NOT HOLE PUNCHING in its native form.

In the BEHAVE WG meeting(last IETF), several people talked of ICE as the new
application traversal approach blessed by IETF, and proposed dropping the
original draft ford-behave-app draft that proposes HOLE PUNCHING TECHNIQUE. As
a result, the application milestone is dropped from the WG charter and
ford-behave-app is not adapted as work group item. This, to me, is pitting ICE
against the basic HOLE PUNCHING TECHNIQUE. This is what I believe to be WRONG.

regards,
suresh


> But perhaps you are referring to something else?
> - Philip
> 
> 
> On 18-Jul-07, at 11:37 , Pyda Srisuresh wrote:
> 
> > Thanks to Henry and Frank for so eloquently articulating this.
> >
> > As both Henry and Frank mentioned, I fully support standardizing on  
> > HOLE
> > PUNCHING TECHNIQUE and leveraging the IETF’s considerable influence  
> > to force
> > NAT vendors to migrate towards implementations that are friendly to  
> > Hole
> > Punching and the Application developers to design applications  
> > using the hole
> > punching techniques. That should be the FIRST order of priority.
> >
> > Yet, what is happening in BEHAVE WG is the other way around. Some  
> > people have
> > put their weight so heavily in favor of ICE that they asked to drop  
> > the work on
> > hole punching technique as WG item. THIS IS WRONG. I no longer see  
> > "NAT
> > Traveral application design guidelines" as BEHAVE WG milestone.
> >
> > Magnus - Can you explain why the IETF is not allowing HOLE PUNCHING  
> > TECHNIQUES
> > to move forward as a WG item?
> >
> > regards,
> > suresh
> >
> > --- "Frank W. Miller" <fwmiller@cornfed.com> wrote:
> >
> >>
> >>
> >> Greetings,
> >>
> >>
> >>
> >> What I think Henry is advocating is a systematic approach to  
> >> making sure
> >> that the content of the ICE spec is right for its intended  
> >> purpose, i.e. a
> >> complete, general NAT traversal solution.  NAT traversal has been  
> >> in my
> >> opinion one of the most difficult aspects of SIP implementation  
> >> and its very
> >> important that its eventual standardized solution be right.  I  
> >> know I spent
> >> a lot of time implementing all the "tests" in the first 3489, only to
> >> discover that they were not that useful.  There are two valid  
> >> technical
> >> concerns that are put forth in this email:
> >>
> >>
> >>
> >> 1)     Viability of correct operation in practical  
> >> implementations.  ICE can
> >> result in very complicated message transfers.
> >>
> >> 2)     Performance.  As pointed out in this email and in other  
> >> places, the
> >> additional latency associated with ICE operations can be significant.
> >>
> >>
> >>
> >> Henry also points out that there are competing approaches (e.g. Hole
> >> Punching) that can perform well in many situations and have better
> >> performance.  IMHO, another approach by the IETF in general would  
> >> be to
> >> standardize on Hole Punching and leverage the IETF's considerable  
> >> influence
> >> to force NAT vendors to migrate towards implementations that are  
> >> friendly to
> >> Hole Punching.
> >>
> >>
> >>
> >> The bottom line is that I agree with Henry in that labeling ICE as an
> >> experimental draft is the right thing to do.  It will allow those
> >> implementing ICE (including myself in all likelihood) to proceed  
> >> but signals
> >> to the users the correct status of the standard's contents.
> >>
> >>
> >>
> >> I also want to applaud loudly the underlying tone of Henry's  
> >> thread here.
> >> That is, in solving specific technical problems within the IETF  
> >> forum,
> >> focusing on empirical observation of working implementations  
> >> should be the
> >> MO that we all strive for when developing and proposing what may  
> >> become
> >> protocol standards.
> >>
> >>
> >>
> >> Thanks,
> >>
> >> FM
> >>
> >>
> >>
> >>
> >>
> >>   _____
> >>
> >> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> >> Sent: Wednesday, July 18, 2007 8:20 AM
> >> To: Magnus Westerlund
> >> Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
> >> Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> >>
> >>
> >>
> >> Magnus,
> >>
> >>
> >>
> >> I sincerely appreciate your interest and this note!
> >>
> >>
> >>
> >>> I don't quite understand why you are requesting a status change  
> >>> of ICE
> >>
> >>
> >>
> >> However the arguments stated by the WG co-chairs and by you here are
> >> procedural in nature, whereas the facts are some cause for serious  
> >> concern,
> >> and I will just mention a few facts:
> >>
> >>
> >>
> >> RFC 3489 (STUN) was also presented as a NAT traversal solution  
> >> back in 2003,
> >> but <draft-ietf-behave-rfc3489bis-07> now clearly admits that  
> >> "STUN is not a
> >> NAT traversal solution by itself. Rather, it is a tool to be used  
> >> in the
> >> context of a NAT traversal solution. This is an important change  
> >> from the
> >> previous version of this specification (RFC 3489), which presented  
> >> STUN as a
> >> complete solution."
> >>
> >> http://www.ietf.org/internet-drafts/draft-ietf-behave- 
> >> rfc3489bis-07.txt
> >>
> >> My guess is, in plain English, RFC 3489 turned out just NOT to be an
> >> acceptable solution as it was hoped to be.
> >>
> >>
> >>
> >> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the  
> >> ICE call
> >> flow with two endpoints behind two NAT (Fig. 22 on page 38)  
> >> require 66
> >> messages just for the SIP INVITE!
> >>
> >> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat- 
> >> scenarios-07.txt
> >>
> >> What happens if the two endpoints are in different ISPs that have  
> >> NAT of
> >> their own in addition to the ones in Fig. 22?
> >>
> >> What happens if one would try ICE along a P2PSIP route with say  
> >> 5-8 hops?
> >> Would there be hundreds of messages? What is the likelihood for  
> >> failure?
> >>
> >>
> >>
> >> By contrast with such approaches based on hope and faith, the  
> >> BEHAVE WG has
> >> produced work based on real measurements and tools to do the  
> >> measurements,
> >> such as
> >>
> >>
> >>
> >> "NAT Classification Test Results"
> >>
> >>
> >>
> >> "Application Design Guidelines for Traversal through Network Address
> >> Translators"
> >>
> >> The measurements for hole punching are reported in
> >>
> >> "Peer-to-Peer Communication Across Network Address Translators"
> >>
> >> http://www.brynosaurus.com/pub/net/p2pnat/
> >>
> >>
> >>
> >> I am also a believer in ICE and have actually urged Jonathan  
> >> Rosenberg to do
> >> this work and he has given his best. Now that we have ICE-17  
> >> version, it is
> >> in the best interest of the industry and the IETF to make it an  
> >> experimental
> >> RFC, but not yet a standard so that:
> >>
> >>  1. Independent, compliant implementations will be developed,
> >>
> >>  2. Testing in events such as SIPit,
> >>
> >>  3. Reporting the results, such as reported for the "hole  
> >> punching" above.
> >>
> >>
> >>
> >> What happens if indeed hole punching proves to be the far more  
> >> effective
> >> approach?
> >>
> >>
> >>
> >> The industry can ill afford a new series of RFC[ICE]-bis and all  
> >> the years
> >> it takes to move from believing to measuring and proving. I am  
> >> sure IETF
> >> procedures were meant in this spirit (though I am not good in  
> >> producing the
> >> right quotes).
> >>
> >>
> >>
> >> Thanks again for your attention to this concern about ICE.
> >>
> >> Also my apology to all the anguish to the numerous contributors to  
> >> the ICE
> >> related I-Ds, but writing test tools and reporting measurements is  
> >> the right
> >> way to go.
> >>
> >>
> >>
> >> Henry
> >>
> >>
> >>
> >>
> >>
> >> -----Original Message-----
> >> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> >> Sent: Wednesday, July 18, 2007 1:11 AM
> >> To: Henry Sinnreich
> >> Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
> >> Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> >>
> >>
> >>
> >> Henry Sinnreich skrev:
> >>
> >>> Following some discussions and soul searching, the best approach for
> >>
> >>> ICE-17 for LC would be to _go ahead and make it an experimental  
> >>> RFC_.
> >>
> >>>
> >>
> >>> This would support the development of interoperable  
> >>> implementations, the
> >>
> >>> collection of deployment data and also hopefully open source code.
> >>
> >>>
> >>
> >>> Though I have full faith in ICE, following the practices that  
> >>> made the
> >>
> >>> IETF successful take precedent here, I believe.
> >>
> >>>
> >>
> >>> This is a change from the attached that I wrote on 7/16/2007.
> >>
> >>>
> >>
> >>> Note: We can only hope the proliferation of SBC in VoIP service  
> >>> provider
> >>
> >>> networks will not make these concerns and ICE for SIP a moot  
> >>> issue, but
> >>
> >>> this is another topic.
> >>
> >>
> >>
> >> Henry,
> >>
> >>
> >>
> >> I don't quite understand why you are requesting a status change of  
> >> ICE?
> >>
> >> To me it appears that ICE is almost finished spec. It has been
> >>
> >> implemented, used and feedback has been influencing the protocol.  
> >> It is
> >>
> >> definitly one of the better examples of current IETF work that  
> >> actually
> >>
> >> has running code. Sure, the implementations may not be fully  
> >> updated yet
> >>
> >> to the latest draft version. But I think it is safe to say that ICE
> >>
> >> works in the intended core cases. If there are some corner cases  
> >> where
> >>
> >> it fails, that may be. But that is not information we will have until
> >>
> >> really large scale deployement or someone makes very extensive tests.
> >>
> >>
> >>
> >> I think the IETF and the Internet has much more benefit from a  
> >> standards
> >>
> >> track solution than any of the risks existing with ICE of today.
> >>
> >>
> >>
> >> And to be clear, proposed standard is allowed to be published  
> >> without a
> >>
> >> single implementation exists. It might not be good practice, but  
> >> it is
> >>
> >> allowed by our process. And as I said before ICE has had enough
> >>
> >> implementation that I am very confortable in balloting YES for it  
> >> when
> >>
> >> it appears on the IESGs table.
> >>
> >>
> >>
> >> Cheers
> >>
> >>
> >>
> >> Magnus Westerlund
> >>
> >>
> >>
> >> IETF Transport Area Director & TSVWG Chair
> >>
> >> --------------------------------------------------------------------- 
> >> -
> >>
> >> Multimedia Technologies, Ericsson Research EAB/TVM/M
> >>
> >> --------------------------------------------------------------------- 
> >> -
> >>
> >> Ericsson AB                | Phone +46 8 4048287
> >>
> >> Torshamsgatan 23           | Fax   +46 8 7575550
> >>
> >> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >>
> >> --------------------------------------------------------------------- 
> >> -
> >>
> >>> _______________________________________________
> >> Behave mailing list
> >> Behave@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/behave
> >>
> >
> >
> >
> >
> >
> > _______________________________________________
> > Behave mailing list
> > Behave@ietf.org
> > https://www1.ietf.org/mailman/listinfo/behave
> 
> 





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



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Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
From: Melinda Shore <mshore@cisco.com>
To: David Barrett <dbarrett@quinthar.com>,
	"'Philip Matthews'" <philip_matthews@magma.ca>,
	Pyda Srisuresh <srisuresh@yahoo.com>
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On 7/18/07 2:13 PM, "David Barrett" <dbarrett@quinthar.com> wrote:
> For example, regarding the "hole punching" approach (by which I assume you
> mean "opening a hole and trying to keep it open" versus ICE's "open a hole
> 'on demand' for each connection"), it's been demonstrated to work at large
> scale with at least 90% direct peer connectivity across the internet, today,
> with a much simpler message flow and implementation.

ICE makes me want to pound red-hot nails into my eye sockets.  That said,
90% really isn't nearly good enough for residential deployments.  For
cheap-o residential equipment, the margins are so thin that a support call
or two is sufficient to wipe out the profit on that sale.  Part of the
reason that ICE is the mess it is is that the group has been iteratively
fixing the various failures by adding more tests and probes and messages
and whatnot.  There are certainly a lot of other ways to achieve reliability
but regardless of which one you choose, reliability really does have to be
a priority.  Ancillary to that is that when you fail you should fail in
a way that can be diagnosed with reasonable ease and transparency
(auditability is good!), and I think we all know what happens when
traditional NAT hole-punching fails - stuff just doesn't work.

Melinda
 


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From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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Thanks to Henry and Frank for so eloquently articulating this. 

As both Henry and Frank mentioned, I fully support standardizing on HOLE
PUNCHING TECHNIQUE and leveraging the IETF’s considerable influence to force
NAT vendors to migrate towards implementations that are friendly to Hole
Punching and the Application developers to design applications using the hole
punching techniques. That should be the FIRST order of priority. 

Yet, what is happening in BEHAVE WG is the other way around. Some people have
put their weight so heavily in favor of ICE that they asked to drop the work on
hole punching technique as WG item. THIS IS WRONG. I no longer see "NAT
Traveral application design guidelines" as BEHAVE WG milestone.

Magnus - Can you explain why the IETF is not allowing HOLE PUNCHING TECHNIQUES
to move forward as a WG item?
 
regards,
suresh

--- "Frank W. Miller" <fwmiller@cornfed.com> wrote:

>  
> 
> Greetings,
> 
>  
> 
> What I think Henry is advocating is a systematic approach to making sure
> that the content of the ICE spec is right for its intended purpose, i.e. a
> complete, general NAT traversal solution.  NAT traversal has been in my
> opinion one of the most difficult aspects of SIP implementation and its very
> important that its eventual standardized solution be right.  I know I spent
> a lot of time implementing all the "tests" in the first 3489, only to
> discover that they were not that useful.  There are two valid technical
> concerns that are put forth in this email:
> 
>  
> 
> 1)     Viability of correct operation in practical implementations.  ICE can
> result in very complicated message transfers.
> 
> 2)     Performance.  As pointed out in this email and in other places, the
> additional latency associated with ICE operations can be significant.
> 
>  
> 
> Henry also points out that there are competing approaches (e.g. Hole
> Punching) that can perform well in many situations and have better
> performance.  IMHO, another approach by the IETF in general would be to
> standardize on Hole Punching and leverage the IETF's considerable influence
> to force NAT vendors to migrate towards implementations that are friendly to
> Hole Punching.
> 
>  
> 
> The bottom line is that I agree with Henry in that labeling ICE as an
> experimental draft is the right thing to do.  It will allow those
> implementing ICE (including myself in all likelihood) to proceed but signals
> to the users the correct status of the standard's contents.
> 
>  
> 
> I also want to applaud loudly the underlying tone of Henry's thread here.
> That is, in solving specific technical problems within the IETF forum,
> focusing on empirical observation of working implementations should be the
> MO that we all strive for when developing and proposing what may become
> protocol standards.
> 
>  
> 
> Thanks,
> 
> FM
> 
>  
> 
>  
> 
>   _____  
> 
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com] 
> Sent: Wednesday, July 18, 2007 8:20 AM
> To: Magnus Westerlund
> Cc: sip@ietf.org; behave@ietf.org; mmusic@ietf.org
> Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
>  
> 
> Magnus,
> 
>  
> 
> I sincerely appreciate your interest and this note!
> 
>  
> 
> > I don't quite understand why you are requesting a status change of ICE
> 
>  
> 
> However the arguments stated by the WG co-chairs and by you here are
> procedural in nature, whereas the facts are some cause for serious concern,
> and I will just mention a few facts:
> 
>  
> 
> RFC 3489 (STUN) was also presented as a NAT traversal solution back in 2003,
> but <draft-ietf-behave-rfc3489bis-07> now clearly admits that "STUN is not a
> NAT traversal solution by itself. Rather, it is a tool to be used in the
> context of a NAT traversal solution. This is an important change from the
> previous version of this specification (RFC 3489), which presented STUN as a
> complete solution."
> 
> http://www.ietf.org/internet-drafts/draft-ietf-behave-rfc3489bis-07.txt
> 
> My guess is, in plain English, RFC 3489 turned out just NOT to be an
> acceptable solution as it was hoped to be. 
> 
>  
> 
> The excellent I-D <draft-ietf-sipping-nat-scenarios-07> shows the ICE call
> flow with two endpoints behind two NAT (Fig. 22 on page 38) require 66
> messages just for the SIP INVITE!
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-07.txt 
> 
> What happens if the two endpoints are in different ISPs that have NAT of
> their own in addition to the ones in Fig. 22?
> 
> What happens if one would try ICE along a P2PSIP route with say 5-8 hops?
> Would there be hundreds of messages? What is the likelihood for failure?
> 
>  
> 
> By contrast with such approaches based on hope and faith, the BEHAVE WG has
> produced work based on real measurements and tools to do the measurements,
> such as
> 
>  
> 
> "NAT Classification Test Results"
> 
>  
> 
> "Application Design Guidelines for Traversal through Network Address
> Translators" 
> 
> The measurements for hole punching are reported in
> 
> "Peer-to-Peer Communication Across Network Address Translators" 
> 
> http://www.brynosaurus.com/pub/net/p2pnat/
> 
>  
> 
> I am also a believer in ICE and have actually urged Jonathan Rosenberg to do
> this work and he has given his best. Now that we have ICE-17 version, it is
> in the best interest of the industry and the IETF to make it an experimental
> RFC, but not yet a standard so that:
> 
>  1. Independent, compliant implementations will be developed,
> 
>  2. Testing in events such as SIPit,
> 
>  3. Reporting the results, such as reported for the "hole punching" above.
> 
>  
> 
> What happens if indeed hole punching proves to be the far more effective
> approach?
> 
>  
> 
> The industry can ill afford a new series of RFC[ICE]-bis and all the years
> it takes to move from believing to measuring and proving. I am sure IETF
> procedures were meant in this spirit (though I am not good in producing the
> right quotes).
> 
>  
> 
> Thanks again for your attention to this concern about ICE. 
> 
> Also my apology to all the anguish to the numerous contributors to the ICE
> related I-Ds, but writing test tools and reporting measurements is the right
> way to go.
> 
>  
> 
> Henry
> 
>  
> 
>  
> 
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
> Sent: Wednesday, July 18, 2007 1:11 AM
> To: Henry Sinnreich
> Cc: mmusic@ietf.org; sip@ietf.org; behave@ietf.org
> Subject: Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
>  
> 
> Henry Sinnreich skrev:
> 
> > Following some discussions and soul searching, the best approach for
> 
> > ICE-17 for LC would be to _go ahead and make it an experimental RFC_.
> 
> > 
> 
> > This would support the development of interoperable implementations, the
> 
> > collection of deployment data and also hopefully open source code.
> 
> > 
> 
> > Though I have full faith in ICE, following the practices that made the
> 
> > IETF successful take precedent here, I believe.
> 
> > 
> 
> > This is a change from the attached that I wrote on 7/16/2007.
> 
> > 
> 
> > Note: We can only hope the proliferation of SBC in VoIP service provider
> 
> > networks will not make these concerns and ICE for SIP a moot issue, but
> 
> > this is another topic.
> 
>  
> 
> Henry,
> 
>  
> 
> I don't quite understand why you are requesting a status change of ICE? 
> 
> To me it appears that ICE is almost finished spec. It has been 
> 
> implemented, used and feedback has been influencing the protocol. It is 
> 
> definitly one of the better examples of current IETF work that actually 
> 
> has running code. Sure, the implementations may not be fully updated yet 
> 
> to the latest draft version. But I think it is safe to say that ICE 
> 
> works in the intended core cases. If there are some corner cases where 
> 
> it fails, that may be. But that is not information we will have until 
> 
> really large scale deployement or someone makes very extensive tests.
> 
>  
> 
> I think the IETF and the Internet has much more benefit from a standards 
> 
> track solution than any of the risks existing with ICE of today.
> 
>  
> 
> And to be clear, proposed standard is allowed to be published without a 
> 
> single implementation exists. It might not be good practice, but it is 
> 
> allowed by our process. And as I said before ICE has had enough 
> 
> implementation that I am very confortable in balloting YES for it when 
> 
> it appears on the IESGs table.
> 
>  
> 
> Cheers
> 
>  
> 
> Magnus Westerlund
> 
>  
> 
> IETF Transport Area Director & TSVWG Chair
> 
> ----------------------------------------------------------------------
> 
> Multimedia Technologies, Ericsson Research EAB/TVM/M
> 
> ----------------------------------------------------------------------
> 
> Ericsson AB                | Phone +46 8 4048287
> 
> Torshamsgatan 23           | Fax   +46 8 7575550
> 
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> ----------------------------------------------------------------------
> 
> > _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 





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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-10.txt 
Date: Thu, 19 Jul 2007 00:52:48 +0200
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Minor comment on section 4.3 (sending requests): it talks about the UAC 
inserting an 'ob' parameter in its Contact URI, should it do the same for 
Contact URIs in responses over an outbound flow (for consistency reasons 
perhaps; allows the other end to detect outbound usage)?

Regards,
Jeroen

Internet-Drafts@ietf.org wrote:
>> 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 : Managing Client Initiated Connections in the Session
>> Initiation Protocol (SIP) Author(s) : C. Jennings, R. Mahy
>> Filename : draft-ietf-sip-outbound-10.txt
>> Pages : 42
>> Date : 2007-7-12
>>
>> The Session Initiation Protocol (SIP) allows proxy servers to
>>   initiate TCP connections and send asynchronous UDP datagrams to
>>   User Agents in order to deliver requests.  However, many practical
>>   considerations, such as the existence of firewalls and Network
>>   Address Translators (NATs), prevent servers from connecting to User
>>   Agents in this way.  This specification defines behaviors for User
>>   Agents, registrars and proxy servers that allow requests to be
>>   delivered on existing connections established by the User Agent.
>>   It also defines keep alive behaviors needed to keep NAT bindings
>>   open and specifies the usage of multiple connections.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-10.txt
>>
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>> "multipart" MIME messages (i.e. documents which have been split
>> up into multiple messages), so check your local documentation on
>> how to manipulate these messages.
>>
>> Below is the data which will enable a MIME compliant mail reader
>> implementation to automatically retrieve the ASCII version of the
>> Internet-Draft.
>>
>
>
>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip 



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Subject: [Sip] comments on draft-camarillo-sip-body-handling-01
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The draft makes support of multipart a SHOULD in RFC 3261. If our 
objective is to use multipart as a vehicle for extending SIP (e.g., by 
including optional bodies in addition to the mandatory one - a location 
object plus SDP for example), this is much better served by a MUST. 
Otherwise, a UAC will have to send a request first with multipart, and 
if its rejected, retry the request with just the supported parts. Its 
effectively the same as the behavior we'd see if the only way to extend 
SIP was with Require, and not Supported.

This draft also strikes me as something that is in the essential 
corrections category, but its debatable.

For multipart/alternative - I don't understand the requirement to order 
them from least to most 'rich'. Is that a general MIME behavior or are 
we defining it that way? If we want to prioritize, shouldn't we use an 
explicit q-value or something? Using order seems like it will be brittle 
(as it has been in SDP where m-lines were never numbered).

The use case described for multipart/alternative - new SDP mechanisms - 
seems kind of bogus. We've decided not to pursue that. Do we have a REAL 
use case for this?

> The reason for not having such a mechanism is that disposition types
>    are typically supported within a context.  Outside that context, a UA
>    (User Agent) may not support the disposition type. 

I think the same can be send for content-types. A UA may support a 
content-type of application/sdp, but only for one disposition 
('session') and not another ('render'). So, generally I would say that a 
UA never really supports a content-type; it supports a set of tuples of 
{content-type, content-disposition, method}.

> UAs process message bodies and body parts in the following way.  On
>    receiving a SIP message, if a body part is not referenced in any way
>    (e.g., there are no header fields or other body parts with a
>    Content-ID URL pointing to it), the UA processes the body part as
>    indicated by its disposition type and the context in which the body
>    part was received.
> 
>    OPEN ISSUE 4: this means that a UA would need to parse all body parts
>    to find references between them before being able to fully process
>    them.  If we are OK with this, we need to include text here
>    explaining it.

This behavior seems like a bad idea, and I don't understand why it is 
needed. If the CID URL is part of an optional extension, then you make 
the body handling optional, and the UA will know to pick up the body 
part if it supports the extension, otherwise not.


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


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From sip-bounces@ietf.org Wed Jul 18 21:19:16 2007
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Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:

> Phil,
>
>> In my view, ICE _is_ a "Hole Punching Technique".
>
> Yes, you absolutely right, though the devil is in the details.
>
> We cannot go here into a shoot-out between hole punching or other
> mentioned on the list vs. ICE, but there seems to be a huge difference
> in effectiveness and performance.

What precise algorithm are you talking about when you say this?
As far as I am concerned, ICE is simply a well-documented hole  
punching algorithm.
It differs from others that I have seen written up in the following  
ways:
- ICE has the concept of peer reflexive addresses, which other hole- 
punching techniques that I have seen do not;
- ICE has the concept of controlling/controlled endpoint, which gives  
the controller additional flexibility in choosing a path;
- ICE does NOT have the concept of port prediction, which some other  
techniques do (e.g., Saikat's TCP work).

Philip


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From sip-bounces@ietf.org Wed Jul 18 21:48:46 2007
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Date: Thu, 19 Jul 2007 09:47:53 +0800
From: Qian Sun <sunqian@huawei.com>
Subject: RE: [Sip] Comments on draft-ietf-sip-multiple-refer-01.txt
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	'Gonzalo Camarillo' <Gonzalo.Camarillo@ericsson.com>
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Hi, Robert and Gonzalo

IMO, "multiple refer" doesn't need AIB part of Referred-By. As Gonzalo =
said, the relay (e.g. conference server) understand what is going on, it =
must have authenticated the identity of the Referrer. So the Referred-By =
header field and 'recipient-list-history'(the same function as Refer-To) =
are creditable for final REFER-Target.

The triggered SIP message sent by the relay just needs to include =
Referred-By header field (maybe plus recipient-list-history), the =
Referred-By token is not necessary (like the example "7.2. Insecure =
REFER" in rfc3892). The information in Referred-By header field is still =
very useful for final REFER-Target to learn the source of the message.=20

It's not worth writing a separate spec for this, just adding few texts =
to this draft may be OK. Since this draft has mentioned Refer-Sub, why =
not says something about Referred-By?

BTW, there are very few commercial IMS implementations on the planet, =
but we believe the future of IMS must not bad, maybe same to Referred-By =
with AIB.

Cheers,
Qian

-----Original Message-----
From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
Sent: Tuesday, July 17, 2007 10:43 PM
To: Gonzalo Camarillo
Cc: Qian Sun; Miguel Garcia; sip; Aki Niemi; Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com
Subject: Re: [Sip] Re: [Sipping] RE: Comments on =
draft-ietf-sip-multiple-refer-01.txt

As much as I like the Referred-By mechanism, before investing too =20
much time into extending Referred-by for this case, ask yourself =20
seriously
if there is a customer for the work. AFAIK, there are no =20
implementations of the AIB part of Referred-by on the planet (please =20
speak up if you
have one) and I know of no-one who plans to build it. Unless that =20
changes, I think we'd all be better off spending our time working on =20
things
that people will actually use.

RjS


On Jul 17, 2007, at 1:34 AM, Gonzalo Camarillo wrote:

> Hi,
>
> [moving this thread from the SIPPING list to this one (SIP) because =20
> this is a SIP document now.]
>
> You are right. It is not trivial to use Referred-By with multiple =20
> REFER. My suggestion is that, the same way as Referred-By is =20
> specified as an extension of refer, you go ahead specify (if you =20
> are interested) how to use Referred-by with multiple REFER in a =20
> different spec.
>
> The body that the referrer would add to its REFER (which is what =20
> the refer target would get) could be an XML list with the =20
> recipients (those that can be disclosed) of the multiple REFER =20
> (plus the referrer). Something similar to what recipients get after =20
> a relay processes a list with copyControl attributes (draft-ietf-=20
> sipping-capacity-attribute-04.txt)... in any case, since multiple =20
> REFER is only used within a context of a service, you need to think =20
> whether having a general Referred-By mechanism is needed or if such =20
> functionality can be achieved in a different way within the =20
> services multiple REFER is used in (e.g., conferencing).
>
> Cheers,
>
> Gonzalo
>
>
> Qian Sun wrote:
>> Hello, Gonzalo
>> RFC3892 says:
>>    The sipfrag inside a Referred-By token MUST contain copies of the
>>    Refer-To, Referred-By, and Date header fields from the REFER =20
>> request.
>> In multiple refer case, the Refer-To header field includes content =20
>> id, not the REFER-Target' URI. The content id is meaningless to =20
>> the REFER-Targets, so the Referred-By token has to include the URI-=20
>> List.
>> Cheers,
>> Qian
>> -----Original Message-----
>> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>> Sent: Monday, July 16, 2007 4:23 PM
>> To: Qian Sun
>> Cc: Aki.Niemi@nokia.com; Markus.Isomaki@nokia.com; =20
>> miguel.an.garcia@nokia.com; hisham.khartabil@gmail.com; =20
>> sipping@ietf.org
>> Subject: Re: Comments on draft-ietf-sip-multiple-refer-01.txt
>> Hi,
>> thanks for your comments. Answers inline:
>> Qian Sun wrote:
>>  > Some comments on the draft...
>>  >
>>  > 1. I think it might be a good idea to talk something on =20
>> "Referred-By" in this draft. Referred-By mechanism is still very =20
>> useful for the multiple refer case. Maybe Refer-To related to a =20
>> URI List with "bcc" or "anonymize" is a trouble. =
"Referred-BY=E2=80=9D in =20
>> the message header is to identify the initial REFER-Issuer. This =20
>> information may be encrypted together with "refer-to" URI list, =20
>> including the copy control level such as to/cc/bcc to the end =20
>> user, i.e. REFER-Target, where the information will be decrypted =20
>> by the REFER-Target. In this case the REFER-Recipient won=E2=80=99t =
be =20
>> able to get rid of the "bcc" URI from the URI list, and the "bcc" =20
>> URI will be sent to other REFER-Targets along with normal to/cc. =20
>> This will be a problem to destroy the blind cc function.
>>  > However, the REFER-issuer may opt to not include a Referred-By =20
>> token. For example:
>>  >    REFER sip:conf-123@example.com;gruu;opaque=3Dhha9s8d-999a  SIP/ =

>> 2.0
>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>  >    Referred-By: Carol <sip:carol@chicago.example.com>
>>  >    Content-Type: application/resource-lists+xml
>>  >    Content-Disposition: recipient-list
>>  >    Content-Length: 362
>>  >    Content-ID: <cn35t8jf02@example.com>
>>  >
>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>  >            =
xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>>  >      <list>
>>  >        <entry uri=3D"sip:bill@example.com?method=3DBYE" />
>>  >        <entry uri=3D"sip:joe@example.org?method=3DBYE" />
>>  >        <entry uri=3D"sip:ted@example.net?method=3DBYE" />
>>  >      </list>
>>  >    </resource-lists>
>>  >
>>  > When the URI list has no "bcc" or "anonymize" URI, REFER-issuer =20
>> may include the Referred-By token in the REFER body.
>> The way Referred-By works is not affected by what this draft says.
>> Therefore, I do not see the need to talk about it.
>>  >
>>  > 2. Nested refer in multiple refer is very interesting, I =20
>> believe it could be a good idea to enroll a "multiple nested =20
>> refer" example into this draft. Just for an example:
>>  >    REFER sip:list-service@example.com;gruu;opaque=3Dhha9s8d-999a   =

>> SIP/2.0
>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>  >    Content-ID: <cn35t8jf02@example.com>
>>  >
>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>  >            =
xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance">
>>  >      <list>
>>  >        <entry =
uri=3D"sip:bill@example.com?method=3DREFER%3FRefer-To%=20
>> 3Dsip:conf-123%40shenzhen.example.com" />
>>  >        <entry =
uri=3D"sip:tom@example.com?method=3DREFER%3FRefer-To%=20
>> 3Dsip:conf-123%40shenzhen.example.com" />
>>  >      </list>
>>  >    </resource-lists>
>>  > In this case, the REFER-Recipient acts as a Refer List service, =20
>> has the similar function to the MESSAGE List service, i.e. =20
>> receives a REFER with URI-List and distributes some REFERs to each =20
>> address in the URI-List.
>>  > Note that, there is only one final REFER-Target in above =20
>> "nested multiple refer" example, not multiple REFER-Targets. In =20
>> some situations, this convergent feature is useful.
>> The draft says the following:
>>     Additionally, REFER-Recipients SHOULD only accept REFER requests
>>     within the context of an application the server understands =20
>> (e.g., a
>>     conferencing application).
>> this text was added to avoid relays being used as random traffic
>> forwarders. Therefore, since a relay needs to understand what is =20
>> going
>> on (e.g., a multiple REFER to trigger BYEs towards several conference
>> attendees) and currently there is no service that uses nested =20
>> refers, I
>> do not think we need to talk about it.
>>  >
>>  > 3. I guess it might be a typo in Figure 2 "8. 200K OK", maybe =20
>> you meant "200 OK"?
>> yes, thanks for pointing it out. We will fix it.
>> Thanks,
>> Gonzalo
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>
>
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [P2PSIP] Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC
	for RFC
Date: Wed, 18 Jul 2007 21:57:34 -0400
To: Henry Sinnreich <hsinnrei@adobe.com>,
	Magnus Westerlund <magnus.westerlund@ericsson.com>,
	Behave WG <behave@ietf.org>, IETF MMUSIC WG <mmusic@ietf.org>,
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On 18-Jul-07, at 21:19 , Philip Matthews wrote:

>
> On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:
>
>> Phil,
>>
>>> In my view, ICE _is_ a "Hole Punching Technique".
>>
>> Yes, you absolutely right, though the devil is in the details.
>>
>> We cannot go here into a shoot-out between hole punching or other
>> mentioned on the list vs. ICE, but there seems to be a huge  
>> difference
>> in effectiveness and performance.
>
> What precise algorithm are you talking about when you say this?
> As far as I am concerned, ICE is simply a well-documented hole  
> punching algorithm.
> It differs from others that I have seen written up in the following  
> ways:
> - ICE has the concept of peer reflexive addresses, which other hole- 
> punching techniques that I have seen do not;
> - ICE has the concept of controlling/controlled endpoint, which  
> gives the controller additional flexibility in choosing a path;
> - ICE does NOT have the concept of port prediction, which some  
> other techniques do (e.g., Saikat's TCP work).

I should add that ICE also handles:
- the case of multi-homed peers,
- peers that want to exchange multiple media streams,
- media streams which are composed of multiple components (e.g. RTP  
and RTCP).
These are cases that a solution for SIP needs to handle, but I have  
not seen discussed in other proposals.

- Philip


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Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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First I apologize for the cross post - this is a mmusic draft, I  
would appreciate replies on the mmusic list only.

Few things I am looking for here  ....

1) specific reasons why this work is less appropriate for standards  
track than other work that IETF puts on the standards track. Keep in  
mind the first step of standards track is Proposed Standards which  
requires basically no interoperability, then it could move to Draft  
Standard where interoperability is shown and finally Full Standard if  
it turns out to be a good idea.  If you are going to post a reply to  
this point, make sure you have read RFC 2026.

2) I am interested in the specifics of problems that ICE fails to  
solve that the WG has not previous considered. There are lots of  
people that have implemented versions of it and think it is a good  
idea - if people are having some problems with it, I'd love to hear  
about details of things that are wrong. That it has a lot of messages  
in some use cases has been known for a long time. It used to have far  
more messages  - engineering was done to consider what was a  
reasonable amount for call set up times and a new design was done to  
bring the message to a level most people invovled with the design  
felt was reasonable. If there are specific other than these that are  
wrong and not considered in the design, now would be a good time to  
get them on the table.

3) If people think that Skype for example has a simpler solution that  
works better than ICE. Feel free to describe exactly what it is.  
Several people invovled with the ICE design have looked in extreme  
detail at Skype and others and so far none of them have come back  
with what we should do different than ICE.

4) ICE can be used in a mode where it only provided the reflexive  
candidate. People were pretty clear that this was far inferior to the  
modes where there are other candidates. If someone would like to  
provide exactly how hole punching works better than this very limited  
ICE technique, I'd be glad to hear the details. This has certainly  
been an topic that any could have brought up over the last several  
years. If someone has specifics, I suggest they get out a full  
proposal really soon. I am very doubtful that hole punching solution  
is fundamentally different than the type of hole punching ICE does.

5) Some people have suggested that if all NATs were behave complaint,  
then we would not need ICE. I strongly disagree. I think the only  
part of ICE we would not need is the TURN relays which are more or  
less an optional part. Again, specifics on what can be removed are  
welcome.

ICE is far more complicated that anyone wants - the problem is, no  
one has come up with a proposal that looks like it comes anywhere  
close to meeting the requirements and if simpler. The design space  
has been extensively explored by many people. The raw data to  
understand what will work on what NATs have been painfully tested and  
documented in publications such as:

http://nutss.gforge.cis.cornell.edu/pub/imc05-tcpnat.pdf

http://tools.ietf.org/id/draft-jennings-behave-test-results-04.txt

https://www.guha.cc/saikat/stunt-results.php?

(Ford used to have a page too but it seems to be gone)

I have a very open mind to considering better ways to do ICE - but I  
need specific suggestions or it is impossible for me to really think  
about if the suggestions represent a better path or not.

I would love to see more open source ICE code (there is lots already)  
- I feel that this is an area where ICE is likely just table stakes  
and owning an proprietary ICE stack will not turn out to be a  
competitive advantage. Having open source implementations where the  
development cost was shared and everyone could get working products  
to market faster would be an advantage to companies. Clearly I expect  
the bulk of ICE implementations to not be open source but I still  
would be keen on doing what I can to contribute to an open source  
version as personally I feel that is in the interest of end users, my  
employer, and the IETF as a SDO.

Cullen <with my individual hat on>



On Jul 18, 2007, at 9:12 AM, Dean Willis wrote:

>
> On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:
>
>> Henry Sinnreich skrev:
>>> Following some discussions and soul searching, the best approach for
>>> ICE-17 for LC would be to _go ahead and make it an experimental  
>>> RFC_.
>>> This would support the development of interoperable  
>>> implementations, the
>>> collection of deployment data and also hopefully open source code.
>>> Though I have full faith in ICE, following the practices that  
>>> made the
>>> IETF successful take precedent here, I believe.
>>> This is a change from the attached that I wrote on 7/16/2007.
>>> Note: We can only hope the proliferation of SBC in VoIP service  
>>> provider
>>> networks will not make these concerns and ICE for SIP a moot  
>>> issue, but
>>> this is another topic.
>>
>> Henry,
>>
>> I don't quite understand why you are requesting a status change of  
>> ICE? To me it appears that ICE is almost finished spec. It has  
>> been implemented, used and feedback has been influencing the  
>> protocol. It is definitly one of the better examples of current  
>> IETF work that actually has running code. Sure, the  
>> implementations may not be fully updated yet to the latest draft  
>> version. But I think it is safe to say that ICE works in the  
>> intended core cases. If there are some corner cases where it  
>> fails, that may be. But that is not information we will have until  
>> really large scale deployement or someone makes very extensive tests.
>
> Henry has, I believe, been very consistent in urging the IETF to  
> have more "running code" experience on every specification (not  
> just ICE) than our process requires. He has, IIRC, also advocated  
> changing the process to require running code before publishing an  
> RFC. He's even suggested at times that one should have running code  
> before submitting a protocol draft.
>
> Another point to consider -- I've known Henry to bark at a tree  
> with no squirrel in it occasionally, but most of the time he's  
> pretty reliable, more so than most technology predictors. If he's  
> worried about ICE, it's probably because he's been talking to  
> people who are trying to implement it and they're telling him it is  
> difficult or that they're having performance or reliability issues.  
> I imagine that there are plenty of internal development teams in  
> various companies that are working on this and not necessarily  
> bringing their results back to the IETF. After all, if everybody  
> sees the emperor as fully clothed, they'd be embarrassed to  
> proclaim otherwise. Of course, I wish all of these developers were  
> tied into the IETF, but then I'm sure I don't want to see every ice- 
> implementor's question on the SIP list. That is why we have a sip- 
> implementors mailing list, right?
>
> Personally, I think ICE is the best solution we have. We KNOW of  
> cases where hole-punching doesn't work, and I think ICE will  
> produce usable results in most of those cases. But that's just my  
> personal opinion from reading the specs. I haven't implemented ICE,  
> and I haven't used anybody else's implementation. Sure, it's  
> complex, but that's because the underlying problem is complex --  
> there are just too many different behaviors out there in "NAT and  
> firewall land". Of course, I could be wrong, too. It's happened  
> before, once or twice ;-). Just how sure are we that we're right  
> about this?
>
> Are we designing a "spruce goose" or a practical system? Only  
> experience will tell us. But some good flight reports from the  
> prototypes could go a long way towards reassuring us of the  
> design's utility.
>
> So far we've had a couple of people report hearsay evidence -- that  
> somebody else has an ICE implementation and it works. Any of you  
> folks have personal experience you can share? Have there been  
> prototypes exercised at the interop events?
>
> --
> Dean
>
>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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On Wed, 2007-07-18 at 21:19 -0400, Philip Matthews wrote:
> On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:
> 
> > Phil,
> >
> >> In my view, ICE _is_ a "Hole Punching Technique".
> >
> > Yes, you absolutely right, though the devil is in the details.
> >
> > We cannot go here into a shoot-out between hole punching or other
> > mentioned on the list vs. ICE, but there seems to be a huge difference
> > in effectiveness and performance.
> 
> What precise algorithm are you talking about when you say this?

Well, here's the algorithm I use with my UA (the most recent versions I
might add):

                            STUN
UA         Registrar       Server
 |  REGISTER   |             |
 |-5060------->|             |
 |  w/rport    |             |
 |             |             |
 |   200 OK    |             |
 |<5060--------|             |
 |             |             |
 |             |             |
 |  STUN BReq  |             |
 |-5060--------------------->|
 |  STUN BReq  |             |
 |-5004--------------------->|
 |  STUN BResp |             |
 |<5060----------------------|
 |  STUN BResp |             |
 |<5004----------------------|
 |             |             |
 |      .      |             |
 |      .      |             |
 |    10sec    |             |
 |      .      |             |
 |      .      |             |
 |             |             |
 |  STUN BReq  |             |
 |-5060--------------------->|
 |  STUN BResp |             |
 |<5060----------------------|
 |  STUN BReq  |             |
 |-5004--------------------->|
 |  STUN BResp |             |
 |<5004----------------------|
 |             |             |
 |             |             |


I pick up the rport addressing during the REGISTER transaction because
of the way my multithreading is done and because its nice to see the
same value coming from two different sources out on the Internet.  The
port numbers on the UA are local port numbers.  STUN BReq is a STUN
binding request and STUN BResp is a STUN binding response.  Port 5004 is
for a single RTP session but the same algorithm is usable for any and
all UDP ports you need, e.g. additional session ports and RTCP.  The
difference in order in the first and second sequence is intentional, the
implementation doesn't care when the responses show up.

This algorithm lets the UA know what its "visible" IP address and port
is all the time (assuming that rport and STUN are working) and keeps it
the same assuming that the NAT doesn't change it unilaterally (which I
have yet to see BTW).  When signaling occurs, the "visible" IP address
and port are used in the Contact headers for SIP and in the SDP for
offers/answers for RTP.

Its just about as simple as it gets and I know you're going to say that
it doesn't work 100% of the time.  I haven't measured how well its done
in deployments I've done as yet.  I guess I need to do something along
those lines.  The good news is, I haven't had anyone complain about it
so I guess I've been lucky.  So let me have it...  ;)

Thanks,
FM







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From sip-bounces@ietf.org Thu Jul 19 00:22:36 2007
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From: Dale.Worley@comcast.net
To: sip@ietf.org
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	(sunqian@huawei.com)
Subject: Re: [Sip] Comments on draft-ietf-sip-subnot-etags-00
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Technical issues:

What if both Suppress-Body-If-Match and SuppressN-Notify-If-Match are
specified?  We probably don't really care, but the document should
make it clear if the combination is standardized or not.

The entity-tag values for a single resource are the same across
subscriptions, but it is not clear whether they are in some sense the
same between subscriptions for different resources, or for different
resource-IDs that refer to "the same" underlying resource.  This may
have significance for renewed subscriptions whose initial requests are
sent to the same resource AOR but which are ultimately routed to
different but "equivalent" resource-IDs.

Nits:

Section 2.2

   o  Relatively low rate of actual notifications triggered by actual
      state changes

Better phrased "Relatively low rate of notifications triggered by
state changes".

Section 3:

The entity-tag header is "SIP-ETag", but the "SIP-" portion is
redundant.  I suppose it's too late to change it to just "ETag".

Sections 3 and 4.2:

"two types of Xs" is better phrased "two Xs".

Section 4.5:

   2.  If the condition evaluates to true, the notifier sends a 204 (No
       Notification) response sends no NOTIFY request.

Should be "sends a 204 (No Notification) response and sends no NOTIFY
request.

Section 5.1:

   The notifier is free to decide the means for generating an
   entity-tag, except for the special "*" value.

Better phrased "free to decide how to generate the entity-tag values,
except that no entity-tag may be the special "*" value."

Section 5.3:

   the notifier MUST suppress the resulting NOTIFY request, and
   generate a 204 (No Notification) "No Notification" response.

Should be "a 204 (No Notification) response".

Sections 8.3 and 8.4:

   This document registers two new SIP header field names.  These
   headers are defined by the following information, which has been
   added to the header fields sub-registry under
   http://www.iana.org/assignments/sip-parameters.

This paragraph is duplicated, and it appears to be addressing both
header registrations together, rather than either one alone.

Dale


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From sip-bounces@ietf.org Thu Jul 19 01:53:58 2007
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Date: Thu, 19 Jul 2007 13:52:48 +0800
From: Linyi Tian <tianlinyi@huawei.com>
Subject: RE: [Sip] Comments and questions
	todraft-ietf-sip-uri-list-conferencing-01.txt
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Hi, Gonzalo
But my question is when and how the rules are provided? For conferencing with pre-defined group the rules are typically specified by group owner (user). How about ad hoc conference?
I think currently in PoC ad hoc session, the rules are only local policy for PoC Server, but not provided by initiator, right? If there is a mean for initiator to manipulate rules after session is created, it seems there is a risk after conf is created but rules are not ready. Some attendees may do something bad which is out of control during that period. This is not expected by the initiator.
BR,
Linyi

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
Sent: Tuesday, July 17, 2007 3:54 PM
To: Linyi Tian
Cc: sip@ietf.org
Subject: Re: [Sip] Comments and questions todraft-ietf-sip-uri-list-conferencing-01.txt

Hi Linyi,

> (1) On section "5.  Conference Server Procedures"
> Many conference servers have restriction to maximum number of attendees 
> for conference. If the users of 'recipient-list' exceeds the limitation, 
> what status code should be returned?

I would return a 403 (Too many recipients).

> ---------------------------------------
> 7.  Security Considerations
>    This document discusses setup of SIP conferences using a 
> request-contained URI-list.  Both conferencing and URI-lists services 
> have specific security requirements which will be summarized here. 
> Conferences generally have authorization rules about who can or cannot 
> join a conference, what type of media can or cannot be used, etc.  This 
> information is used by the focus to admit or deny participation in a 
> conference.  It is RECOMMENDED that these types of authorization rules 
> be used to provide security for a SIP conference.
> ----------------------------------------
> (2) When and how the authorization rules are specified and applied to 
> the conference? Before the conference creation seems not possible since 
> Conf-URI is not assigned. If specified after the conference creation, it 
> seems there is a risk after conf is created but rules are not ready. 
> Some attendees may do something bad which is out of control during that 
> period.

They are applied at conference creation. The server receives the 
request, creates the conference with its rules, and only adds the 
participants in the list if those rules allow it to do it. This is 
already implemented in the OMA PoC (Push-to-talk over Cellular) service.

Thanks for your comments,

Gonzalo




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From sip-bounces@ietf.org Thu Jul 19 02:14:49 2007
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Subject: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
From: jh@tutpro.com (Juha Heinanen)
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a while back i read ice tutorial slides and my impression was that the
protocol was very complex (pity implementers) and provided only a bit
better nat traversal capability than what [open]ser/rtpproxy/mediaproxy
have been offering to lots of users for years already.

because of this (complexity, little added value), i doubt that full ice
will be widely implemented or used.  i don't have any opinion on if ietf
should publish the spec or not though, because there are already so many
irrelevant rfcs out there that one more doesn't make any difference.

-- juha



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From sip-bounces@ietf.org Thu Jul 19 02:45:06 2007
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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: mmusic@ietf.org, "ext Juha Heinanen" <jh@tutpro.com>
Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
Date: Thu, 19 Jul 2007 09:43:17 +0300
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[As Cullen pointed out, this is better located on MMUSIC. Thanks.]

On Thursday 19 July 2007 09:14:29 ext Juha Heinanen wrote:
> a while back i read ice tutorial slides and my impression was that the
> protocol was very complex (pity implementers) and provided only a bit
> better nat traversal capability than what [open]ser/rtpproxy/mediaproxy
> have been offering to lots of users for years already.

Media proxying is totally broken for a bunch of different reasons:

1/ Like many SIP ALGs - it screws the signaling when it does not understand=
=20
something. Last time I tried OpenSER media proxying, TCP/COMEDIA sessions d=
id=20
not work at all (even with public unfirewalled IPs on both endpoints).

2/ At least mediaproxy implementation does not scale at all. It might be go=
od=20
if one's business consists of selling relay boxes. That's an implementation=
=20
issue however.

3/ It is totally fatally insecure by design. It is trivial to DoS, or worse=
,=20
hijack, a media flow by sending fake RTP probe packets to the relay.

> because of this (complexity, little added value), i doubt that full ice
> will be widely implemented or used.  i don't have any opinion on if ietf
> should publish the spec or not though, because there are already so many
> irrelevant rfcs out there that one more doesn't make any difference.

3/ ICE is secure: no voice hammer, no media hijacking.

2/ ICE scales well as it puts the hard work on the endpoints (and usage of=
=20
TURN may be limited to worst cases).

1/ ICE does not screw up the SIP signaling, and you do not need to buy a SI=
P=20
server upgrade to support the next SDP extension.

In all due fairness, simple hole punching also has advantages 1 and 2 over=
=20
media relaying. And it is about as secure as SIP without any traversal=20
mechanism, which is to say, less secure than ICE, but much more secure than=
=20
media relaying.


Regards,

=2D-=20
R=E9mi Denis-Courmont


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To: Qian Sun <sunqian@huawei.com>
Subject: Re: [Sip] Comments on draft-ietf-sip-multiple-refer-01.txt
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Hi,

as I stated before, at this point, if you think specifying this feature=20
is useful, it should be done in a separate (new) spec. It is too late in=20
the process to add new features to the multiple-refer draft.

Cheers,

Gonzalo


Qian Sun wrote:
> Hi, Robert and Gonzalo
>=20
> IMO, "multiple refer" doesn't need AIB part of Referred-By. As Gonzalo =
said, the relay (e.g. conference server) understand what is going on, it =
must have authenticated the identity of the Referrer. So the Referred-By =
header field and 'recipient-list-history'(the same function as Refer-To) =
are creditable for final REFER-Target.
>=20
> The triggered SIP message sent by the relay just needs to include Refer=
red-By header field (maybe plus recipient-list-history), the Referred-By =
token is not necessary (like the example "7.2. Insecure REFER" in rfc3892=
). The information in Referred-By header field is still very useful for f=
inal REFER-Target to learn the source of the message.=20
>=20
> It's not worth writing a separate spec for this, just adding few texts =
to this draft may be OK. Since this draft has mentioned Refer-Sub, why no=
t says something about Referred-By?
>=20
> BTW, there are very few commercial IMS implementations on the planet, b=
ut we believe the future of IMS must not bad, maybe same to Referred-By w=
ith AIB.
>=20
> Cheers,
> Qian
>=20
> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Tuesday, July 17, 2007 10:43 PM
> To: Gonzalo Camarillo
> Cc: Qian Sun; Miguel Garcia; sip; Aki Niemi; Markus.Isomaki@nokia.com> =
<Markus.Isomaki@nokia.com
> Subject: Re: [Sip] Re: [Sipping] RE: Comments on draft-ietf-sip-multipl=
e-refer-01.txt
>=20
> As much as I like the Referred-By mechanism, before investing too =20
> much time into extending Referred-by for this case, ask yourself =20
> seriously
> if there is a customer for the work. AFAIK, there are no =20
> implementations of the AIB part of Referred-by on the planet (please =20
> speak up if you
> have one) and I know of no-one who plans to build it. Unless that =20
> changes, I think we'd all be better off spending our time working on =20
> things
> that people will actually use.
>=20
> RjS
>=20
>=20
> On Jul 17, 2007, at 1:34 AM, Gonzalo Camarillo wrote:
>=20
>> Hi,
>>
>> [moving this thread from the SIPPING list to this one (SIP) because =20
>> this is a SIP document now.]
>>
>> You are right. It is not trivial to use Referred-By with multiple =20
>> REFER. My suggestion is that, the same way as Referred-By is =20
>> specified as an extension of refer, you go ahead specify (if you =20
>> are interested) how to use Referred-by with multiple REFER in a =20
>> different spec.
>>
>> The body that the referrer would add to its REFER (which is what =20
>> the refer target would get) could be an XML list with the =20
>> recipients (those that can be disclosed) of the multiple REFER =20
>> (plus the referrer). Something similar to what recipients get after =20
>> a relay processes a list with copyControl attributes (draft-ietf-=20
>> sipping-capacity-attribute-04.txt)... in any case, since multiple =20
>> REFER is only used within a context of a service, you need to think =20
>> whether having a general Referred-By mechanism is needed or if such =20
>> functionality can be achieved in a different way within the =20
>> services multiple REFER is used in (e.g., conferencing).
>>
>> Cheers,
>>
>> Gonzalo
>>
>>
>> Qian Sun wrote:
>>> Hello, Gonzalo
>>> RFC3892 says:
>>>    The sipfrag inside a Referred-By token MUST contain copies of the
>>>    Refer-To, Referred-By, and Date header fields from the REFER =20
>>> request.
>>> In multiple refer case, the Refer-To header field includes content =20
>>> id, not the REFER-Target' URI. The content id is meaningless to =20
>>> the REFER-Targets, so the Referred-By token has to include the URI-=20
>>> List.
>>> Cheers,
>>> Qian
>>> -----Original Message-----
>>> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>>> Sent: Monday, July 16, 2007 4:23 PM
>>> To: Qian Sun
>>> Cc: Aki.Niemi@nokia.com; Markus.Isomaki@nokia.com; =20
>>> miguel.an.garcia@nokia.com; hisham.khartabil@gmail.com; =20
>>> sipping@ietf.org
>>> Subject: Re: Comments on draft-ietf-sip-multiple-refer-01.txt
>>> Hi,
>>> thanks for your comments. Answers inline:
>>> Qian Sun wrote:
>>>  > Some comments on the draft...
>>>  >
>>>  > 1. I think it might be a good idea to talk something on =20
>>> "Referred-By" in this draft. Referred-By mechanism is still very =20
>>> useful for the multiple refer case. Maybe Refer-To related to a =20
>>> URI List with "bcc" or "anonymize" is a trouble. "Referred-BY=E2=80=9D=
 in =20
>>> the message header is to identify the initial REFER-Issuer. This =20
>>> information may be encrypted together with "refer-to" URI list, =20
>>> including the copy control level such as to/cc/bcc to the end =20
>>> user, i.e. REFER-Target, where the information will be decrypted =20
>>> by the REFER-Target. In this case the REFER-Recipient won=E2=80=99t b=
e =20
>>> able to get rid of the "bcc" URI from the URI list, and the "bcc" =20
>>> URI will be sent to other REFER-Targets along with normal to/cc. =20
>>> This will be a problem to destroy the blind cc function.
>>>  > However, the REFER-issuer may opt to not include a Referred-By =20
>>> token. For example:
>>>  >    REFER sip:conf-123@example.com;gruu;opaque=3Dhha9s8d-999a  SIP/=
=20
>>> 2.0
>>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>>  >    Referred-By: Carol <sip:carol@chicago.example.com>
>>>  >    Content-Type: application/resource-lists+xml
>>>  >    Content-Disposition: recipient-list
>>>  >    Content-Length: 362
>>>  >    Content-ID: <cn35t8jf02@example.com>
>>>  >
>>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance"=
>
>>>  >      <list>
>>>  >        <entry uri=3D"sip:bill@example.com?method=3DBYE" />
>>>  >        <entry uri=3D"sip:joe@example.org?method=3DBYE" />
>>>  >        <entry uri=3D"sip:ted@example.net?method=3DBYE" />
>>>  >      </list>
>>>  >    </resource-lists>
>>>  >
>>>  > When the URI list has no "bcc" or "anonymize" URI, REFER-issuer =20
>>> may include the Referred-By token in the REFER body.
>>> The way Referred-By works is not affected by what this draft says.
>>> Therefore, I do not see the need to talk about it.
>>>  >
>>>  > 2. Nested refer in multiple refer is very interesting, I =20
>>> believe it could be a good idea to enroll a "multiple nested =20
>>> refer" example into this draft. Just for an example:
>>>  >    REFER sip:list-service@example.com;gruu;opaque=3Dhha9s8d-999a  =
=20
>>> SIP/2.0
>>>  >    Refer-To: <cid:cn35t8jf02@example.com>
>>>  >    Content-ID: <cn35t8jf02@example.com>
>>>  >
>>>  >    <?xml version=3D"1.0" encoding=3D"UTF-8"?>
>>>  >    <resource-lists xmlns=3D"urn:ietf:params:xml:ns:resource-lists"
>>>  >            xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance"=
>
>>>  >      <list>
>>>  >        <entry uri=3D"sip:bill@example.com?method=3DREFER%3FRefer-T=
o%=20
>>> 3Dsip:conf-123%40shenzhen.example.com" />
>>>  >        <entry uri=3D"sip:tom@example.com?method=3DREFER%3FRefer-To=
%=20
>>> 3Dsip:conf-123%40shenzhen.example.com" />
>>>  >      </list>
>>>  >    </resource-lists>
>>>  > In this case, the REFER-Recipient acts as a Refer List service, =20
>>> has the similar function to the MESSAGE List service, i.e. =20
>>> receives a REFER with URI-List and distributes some REFERs to each =20
>>> address in the URI-List.
>>>  > Note that, there is only one final REFER-Target in above =20
>>> "nested multiple refer" example, not multiple REFER-Targets. In =20
>>> some situations, this convergent feature is useful.
>>> The draft says the following:
>>>     Additionally, REFER-Recipients SHOULD only accept REFER requests
>>>     within the context of an application the server understands =20
>>> (e.g., a
>>>     conferencing application).
>>> this text was added to avoid relays being used as random traffic
>>> forwarders. Therefore, since a relay needs to understand what is =20
>>> going
>>> on (e.g., a multiple REFER to trigger BYEs towards several conference
>>> attendees) and currently there is no service that uses nested =20
>>> refers, I
>>> do not think we need to talk about it.
>>>  >
>>>  > 3. I guess it might be a typo in Figure 2 "8. 200K OK", maybe =20
>>> you meant "200 OK"?
>>> yes, thanks for pointing it out. We will fix it.
>>> Thanks,
>>> Gonzalo
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Linyi Tian <tianlinyi@huawei.com>
Subject: Re: [Sip] Comments and questions
	todraft-ietf-sip-uri-list-conferencing-01.txt
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Hi,

the problem of doing many things in one SIP operation (i.e., in a SIP 
request) is that it is difficult to report errors if something goes wrong.

If you want to create an ad-hoc conference with rules that are different 
from the default ad-hoc conference, you can always use a different way 
to create the conference (i.e., without using URI lists; using a 
conference control protocol). URI lists are an optimization for the more 
typical use cases where session establishment should be very fast.

Cheers,

Gonzalo


Linyi Tian wrote:
> Hi, Gonzalo
> But my question is when and how the rules are provided? For conferencing with pre-defined group the rules are typically specified by group owner (user). How about ad hoc conference?
> I think currently in PoC ad hoc session, the rules are only local policy for PoC Server, but not provided by initiator, right? If there is a mean for initiator to manipulate rules after session is created, it seems there is a risk after conf is created but rules are not ready. Some attendees may do something bad which is out of control during that period. This is not expected by the initiator.
> BR,
> Linyi
> 
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Sent: Tuesday, July 17, 2007 3:54 PM
> To: Linyi Tian
> Cc: sip@ietf.org
> Subject: Re: [Sip] Comments and questions todraft-ietf-sip-uri-list-conferencing-01.txt
> 
> Hi Linyi,
> 
>> (1) On section "5.  Conference Server Procedures"
>> Many conference servers have restriction to maximum number of attendees 
>> for conference. If the users of 'recipient-list' exceeds the limitation, 
>> what status code should be returned?
> 
> I would return a 403 (Too many recipients).
> 
>> ---------------------------------------
>> 7.  Security Considerations
>>    This document discusses setup of SIP conferences using a 
>> request-contained URI-list.  Both conferencing and URI-lists services 
>> have specific security requirements which will be summarized here. 
>> Conferences generally have authorization rules about who can or cannot 
>> join a conference, what type of media can or cannot be used, etc.  This 
>> information is used by the focus to admit or deny participation in a 
>> conference.  It is RECOMMENDED that these types of authorization rules 
>> be used to provide security for a SIP conference.
>> ----------------------------------------
>> (2) When and how the authorization rules are specified and applied to 
>> the conference? Before the conference creation seems not possible since 
>> Conf-URI is not assigned. If specified after the conference creation, it 
>> seems there is a risk after conf is created but rules are not ready. 
>> Some attendees may do something bad which is out of control during that 
>> period.
> 
> They are applied at conference creation. The server receives the 
> request, creates the conference with its rules, and only adds the 
> participants in the list if those rules allow it to do it. This is 
> already implemented in the OMA PoC (Push-to-talk over Cellular) service.
> 
> Thanks for your comments,
> 
> Gonzalo
> 
> 



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From: "vishal jogi" <vishaljogi@gmail.com>
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Hi,
I have one doubt.

i m subscribing to RLS uri,
which is refering one Resource-list which doesn't have any <entry>
mean it's blank <LIST>

server is sending 404 as response for Subscribe, can anyone tell me it's
correct behaviour of server,
as that <LIST> is existing

Thanks,
Vishal...

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<div>Hi,</div>
<div>I have one doubt.</div>
<div>&nbsp;</div>
<div>i m subscribing to RLS uri,</div>
<div>which is refering one Resource-list which doesn&#39;t have any &lt;entry&gt;</div>
<div>mean it&#39;s blank &lt;LIST&gt;</div>
<div>&nbsp;</div>
<div>server is sending 404 as response for Subscribe, can anyone tell me it&#39;s correct behaviour of server, </div>
<div>as that &lt;LIST&gt; is existing</div>
<div>&nbsp;</div>
<div>Thanks,</div>
<div>Vishal...</div>

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Date: Thu, 19 Jul 2007 15:40:23 +0800
From: Linyi Tian <tianlinyi@huawei.com>
Subject: RE: [Sip] Comments and questions
	todraft-ietf-sip-uri-list-conferencing-01.txt
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Hi, Gonzalo

> the problem of doing many things in one SIP operation (i.e., in a SIP 
> request) is that it is difficult to report errors if something goes wrong.
[Linyi] I think this problem might exist in URI List Conferencing. For example, too many recipients, URI format is invalid, etc. Do we need to consider these error conditions?

> If you want to create an ad-hoc conference with rules that are different 
> from the default ad-hoc conference, you can always use a different way 
> to create the conference (i.e., without using URI lists; using a 
> conference control protocol). URI lists are an optimization for the more 
> typical use cases where session establishment should be very fast.
[Linyi] I agree fast is important. Beyond fast experience users also care about rules of a conference. And in some scenarios default rules obviously can not fulfill their expectation. For example, Initiator wants to provide some simple rules such as maximum attendee number, whether allow inviting new person, etc. I have to admit conf control protocol is too heavy to do that for ad hoc conference.

BR,
Linyi

Linyi Tian wrote:
c Hi, Gonzalo
> But my question is when and how the rules are provided? For conferencing with pre-defined group the rules are typically specified by group owner (user). How about ad hoc conference?
> I think currently in PoC ad hoc session, the rules are only local policy for PoC Server, but not provided by initiator, right? If there is a mean for initiator to manipulate rules after session is created, it seems there is a risk after conf is created but rules are not ready. Some attendees may do something bad which is out of control during that period. This is not expected by the initiator.
> BR,
> Linyi
> 
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com] 
> Sent: Tuesday, July 17, 2007 3:54 PM
> To: Linyi Tian
> Cc: sip@ietf.org
> Subject: Re: [Sip] Comments and questions todraft-ietf-sip-uri-list-conferencing-01.txt
> 
> Hi Linyi,
> 
>> (1) On section "5.  Conference Server Procedures"
>> Many conference servers have restriction to maximum number of attendees 
>> for conference. If the users of 'recipient-list' exceeds the limitation, 
>> what status code should be returned?
> 
> I would return a 403 (Too many recipients).
> 
>> ---------------------------------------
>> 7.  Security Considerations
>>    This document discusses setup of SIP conferences using a 
>> request-contained URI-list.  Both conferencing and URI-lists services 
>> have specific security requirements which will be summarized here. 
>> Conferences generally have authorization rules about who can or cannot 
>> join a conference, what type of media can or cannot be used, etc.  This 
>> information is used by the focus to admit or deny participation in a 
>> conference.  It is RECOMMENDED that these types of authorization rules 
>> be used to provide security for a SIP conference.
>> ----------------------------------------
>> (2) When and how the authorization rules are specified and applied to 
>> the conference? Before the conference creation seems not possible since 
>> Conf-URI is not assigned. If specified after the conference creation, it 
>> seems there is a risk after conf is created but rules are not ready. 
>> Some attendees may do something bad which is out of control during that 
>> period.
> 
> They are applied at conference creation. The server receives the 
> request, creates the conference with its rules, and only adds the 
> participants in the list if those rules allow it to do it. This is 
> already implemented in the OMA PoC (Push-to-talk over Cellular) service.
> 
> Thanks for your comments,
> 
> Gonzalo
> 
> 





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Hi,

>> the problem of doing many things in one SIP operation (i.e., in a 
>> SIP request) is that it is difficult to report errors if something 
>> goes wrong.

> [Linyi] I think this problem might exist in URI List Conferencing.
> For example, too many recipients, URI format is invalid, etc. Do we
> need to consider these error conditions?

Yes, it exists. That's why you want to use that mechanism only in the 
context of a well-defined service that is scoped well-enough (e.g., 
ad-hoc sessions in PoC) so that it does not create complex error 
situations when the mechanism is used.

Cheers,

Gonzalo



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From sip-bounces@ietf.org Thu Jul 19 04:55:06 2007
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Date: Thu, 19 Jul 2007 14:24:53 +0530
From: "Vikram Chhibber" <vikram.chhibber@gmail.com>
To: "vishal jogi" <vishaljogi@gmail.com>
Subject: Re: [Sip] Event - list subscribe
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I don't think that this is correct behavior of the RLS. If the RLS
understands the uri pointing to a resource-list. It should accept this
subscription. Even though the list is empty. The user/administrator
may provision entries in this list in future by non-SIP mechanism
(XCAP).

On 7/19/07, vishal jogi <vishaljogi@gmail.com> wrote:
> Hi,
> I have one doubt.
>
> i m subscribing to RLS uri,
> which is refering one Resource-list which doesn't have any <entry>
> mean it's blank <LIST>
>
> server is sending 404 as response for Subscribe, can anyone tell me it's
> correct behaviour of server,
> as that <LIST> is existing
>
> Thanks,
> Vishal...
> _______________________________________________
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> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on
> current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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	handling)
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Hi Paul,

> Thanks for sending out in advance. Maybe we can get into the discussion 
> of the slides now, before the meeting.

yes, that was the intention.

> Regarding references to body parts:
> 
> IMO it is a *bad* idea to make the decision about processing of a body 
> part "on its own" dependent on whether the node doing so has detected a 
> reference to that body part or not.
> 
> For one thing, that requires a multi-pass parsing mechanism to check for 
> references before deciding what processing to do on a particular part. 
> For another, it requires that the recipient understand every reference. 
> For instance, the sender may have included an extension header that 
> contains a reference to a body part. If the extension isn't understood, 
> then the body part should not be processed. But if the recipient doesn't 
> understand the extension then it won't recognize the reference either, 
> and so will attempt to process the body part as one without references.

yes, these two issues are described in the draft.

> The only good solution I see to this is to partition the 
> Content-Disposition and/or Content-Type values whose processing is 
> dependent on reference from those whose processing is not dependent on a 
> reference. For instance, we could simply define a new 
> Content-Disposition of "by-reference".
> 
> Then a normal sequence of processing would be to first process all the 
> body parts according to their type and disposition alone. Any with 
> disposition "by-reference" would be set aside. Then, when processing 
> headers, if a header with a reference is encountered the appropriate 
> processing could be applied to the set aside body part. References 
> within other body parts would be a little more complex, but you get the 
> idea.

yes. When looking for references, one would need to process the header 
fields plus the body parts with a disposition type different than 
'by-reference', since, as you point out, those body parts could contain 
references to 'by-reference' body parts.

The only case we would not cover would be two (or more) 'by-reference' 
body parts referencing each other. But we can just say that this is 
forbidden (no extension does that at present).


> This would not prevent reference to body parts with other dispositions. 
> It just means that those body parts would also be processed according to 
> their type and disposition, in addition to any processing they might get 
> according to their reference. The "by-reference" disposition is just a 
> defintion that gets no implicit processing.

yes, the draft already talks about the possibility of having several 
references to a body part. In such a case, the body part would need to 
be processed several times (in different ways). However, the case you 
are talking about is not currently addresses (i.e., a body part is 
processed once based on a reference and another time based on its 
content disposition). None of these situations occur within current 
extensions, but they could happen in the future.

> An advantage of this approach is that by-reference body parts don't need 
> to be parsed unless the reference is processed and the action required 
> in processing the reference requires the node to parse the referenced 
> body. (Interaction with handling=required in this case is TBD.) Even a 
> multipart with disposition of by-reference might not need to be parsed.

The interaction with handling=required would be that if you are not able 
to find a reference to a 'by-reference' body part with 
handling=required, you report an error because you have not been able to 
process the body part.

This approach would work and has the advantage to be explicit about how 
a body part is expected to be processed... I would like to hear more 
opinions on this.

Cheers,

Gonzalo


> Gonzalo Camarillo wrote:
>> Folks,
>>
>> here you have the slides I have put together for the body (MIME) 
>> handling presentation on Monday:
>>
>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>
>>
>> These slides relate to this draft:
>>
>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>
>>
>> The slides discuss all the open issues that have been brought up in 
>> the list in the last months.
>>
>> Comments are welcome.
>>
>> Thanks,
>>
>> Gonzalo



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From sip-bounces@ietf.org Thu Jul 19 06:10:46 2007
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi Paul,

> Regarding Content-Disposition in multiparts:
> 
> While it seems "wrong" to me, I can live with using "render" as the 
> normal disposition for multipart, *if* we can do a careful job of 
> defining what "render" actually means for SIP. Used this way it clearly 
> doesn't always mean to display the content to a user.

An entity receiving a 'multipart/mixed' body will process its body parts 
based on their disposition types.If the disposition type of a particular 
body part is not understood, it will be displayed to the user because 
the default disposition type is 'render'.

Therefore, the 'multipart/mixed' body will never be processed based on 
its disposition type. This means that it does not make much sense to 
talk about its disposition type as a whole. However, we need to have one 
in order to be able to set its handling parameter. That's why we can 
either keep 'render' explaining that it is actually never used or define 
a new disposition type.

Since existing implementations will not understand our newly defined 
disposition type, they will treat it as 'render' anyway... but the 
resulting behavior would be the same. That's why it may make sense to 
keep using 'render'.

On the other hand, semantically, it would make more sense to define a 
new disposition type.

Also, if we decide to define the 'by-reference' disposition type (per a 
different email thread), it could be used here as well.

Cheers,

Gonzalo



> Gonzalo Camarillo wrote:
>> Folks,
>>
>> here you have the slides I have put together for the body (MIME) 
>> handling presentation on Monday:
>>
>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>
>>
>> These slides relate to this draft:
>>
>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>
>>
>> The slides discuss all the open issues that have been brought up in 
>> the list in the last months.
>>
>> Comments are welcome.
>>
>> Thanks,
>>
>> Gonzalo
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>



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Hi,=20

>>"in order for the mechanism to work, SBC-type-of-entities must permit=20
>>DTLS, TLS, ICE, or HIP messages to be exchanged in the media path."
>>=20
>>A small question for clarification: at what point must this exchange=20
>>(two-way, I assume) work? As soon as the UAS has received the INVITE?
>=20
>Yes.

A little more on this.

As we have discussed before, you may not have two-way connectivity
before 200 OK.

Now assume that you will have two-way connectivity before you have even
sent the SDP answer. I am pretty sure what will not work with most SBCs.

Also, even if you don't have any SBCs, but you still have NATs, the
calling user often needs to send something in order to open the NAT
binding, and allow media plane traffic (including your identity
exchange) from the remote end. If you use ICE, or stand-alone STUN, you
will have an open NAT binding, but if your terminal opens the network
e.g. by sending dummy RTP packets, it will not be able to do so until it
has received the SDP answer.

So, I also think your solution requires support of ICE/STUN, and in the
case of ICE it also requires that the SBCs don't mess up the SDP so that
ICE will not be used.

Regards,

Christer



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From sip-bounces@ietf.org Thu Jul 19 06:26:24 2007
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Subject: RE: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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Hi Cullen, I think you bring up lot of really great and detailed questions.

However, speaking for myself at least, answering those questions doesn't
really address my concern.  All I want to know is:

	Does ICE work, and how well?

What "works" means is entirely open to debate.  I've tossed out "high peer
connectivity" and "low connection setup time" as the most important
criteria, and I've put hard data behind those numbers as a stake in the
ground for comparison, but I'm open to other metrics as well.

But regardless of the metrics, I want somebody who uses it and loves it and
bases their business upon its success to stand up and explain why it's
great.  And I want this person to provide real-world data, sampled on a wide
scale, proving that ICE "works" -- by any definition.

Basically, I want ICE firmly grounded in wide-scale testing and real-world
data, and answering the questions below doesn't bring us any closer to that.
Whether it's level 10 or level 12 doesn't produce new data.  How its
algorithms differ from competing ones doesn't show that they're any better
or worse.  Detailed proposals for change don't shed any light on whether
those changes improve or worsen it, because its current state is completely
unmeasured.

The most powerful thing the IETF can do to standardize the world upon ICE is
prove that it works better than any alternative.  Because until that's
proven, we just need to take it as a matter of faith.

-david

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Wednesday, July 18, 2007 7:56 PM
> To: SIP; Behave WG; mmusic@ietf.org
> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> 
> First I apologize for the cross post - this is a mmusic draft, I
> would appreciate replies on the mmusic list only.
> 
> Few things I am looking for here  ....
> 
> 1) specific reasons why this work is less appropriate for standards
> track than other work that IETF puts on the standards track. Keep in
> mind the first step of standards track is Proposed Standards which
> requires basically no interoperability, then it could move to Draft
> Standard where interoperability is shown and finally Full Standard if
> it turns out to be a good idea.  If you are going to post a reply to
> this point, make sure you have read RFC 2026.
> 
> 2) I am interested in the specifics of problems that ICE fails to
> solve that the WG has not previous considered. There are lots of
> people that have implemented versions of it and think it is a good
> idea - if people are having some problems with it, I'd love to hear
> about details of things that are wrong. That it has a lot of messages
> in some use cases has been known for a long time. It used to have far
> more messages  - engineering was done to consider what was a
> reasonable amount for call set up times and a new design was done to
> bring the message to a level most people invovled with the design
> felt was reasonable. If there are specific other than these that are
> wrong and not considered in the design, now would be a good time to
> get them on the table.
> 
> 3) If people think that Skype for example has a simpler solution that
> works better than ICE. Feel free to describe exactly what it is.
> Several people invovled with the ICE design have looked in extreme
> detail at Skype and others and so far none of them have come back
> with what we should do different than ICE.
> 
> 4) ICE can be used in a mode where it only provided the reflexive
> candidate. People were pretty clear that this was far inferior to the
> modes where there are other candidates. If someone would like to
> provide exactly how hole punching works better than this very limited
> ICE technique, I'd be glad to hear the details. This has certainly
> been an topic that any could have brought up over the last several
> years. If someone has specifics, I suggest they get out a full
> proposal really soon. I am very doubtful that hole punching solution
> is fundamentally different than the type of hole punching ICE does.
> 
> 5) Some people have suggested that if all NATs were behave complaint,
> then we would not need ICE. I strongly disagree. I think the only
> part of ICE we would not need is the TURN relays which are more or
> less an optional part. Again, specifics on what can be removed are
> welcome.
> 
> ICE is far more complicated that anyone wants - the problem is, no
> one has come up with a proposal that looks like it comes anywhere
> close to meeting the requirements and if simpler. The design space
> has been extensively explored by many people. The raw data to
> understand what will work on what NATs have been painfully tested and
> documented in publications such as:
> 
> http://nutss.gforge.cis.cornell.edu/pub/imc05-tcpnat.pdf
> 
> http://tools.ietf.org/id/draft-jennings-behave-test-results-04.txt
> 
> https://www.guha.cc/saikat/stunt-results.php?
> 
> (Ford used to have a page too but it seems to be gone)
> 
> I have a very open mind to considering better ways to do ICE - but I
> need specific suggestions or it is impossible for me to really think
> about if the suggestions represent a better path or not.
> 
> I would love to see more open source ICE code (there is lots already)
> - I feel that this is an area where ICE is likely just table stakes
> and owning an proprietary ICE stack will not turn out to be a
> competitive advantage. Having open source implementations where the
> development cost was shared and everyone could get working products
> to market faster would be an advantage to companies. Clearly I expect
> the bulk of ICE implementations to not be open source but I still
> would be keen on doing what I can to contribute to an open source
> version as personally I feel that is in the interest of end users, my
> employer, and the IETF as a SDO.
> 
> Cullen <with my individual hat on>
> 
> 
> 
> On Jul 18, 2007, at 9:12 AM, Dean Willis wrote:
> 
> >
> > On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:
> >
> >> Henry Sinnreich skrev:
> >>> Following some discussions and soul searching, the best approach for
> >>> ICE-17 for LC would be to _go ahead and make it an experimental
> >>> RFC_.
> >>> This would support the development of interoperable
> >>> implementations, the
> >>> collection of deployment data and also hopefully open source code.
> >>> Though I have full faith in ICE, following the practices that
> >>> made the
> >>> IETF successful take precedent here, I believe.
> >>> This is a change from the attached that I wrote on 7/16/2007.
> >>> Note: We can only hope the proliferation of SBC in VoIP service
> >>> provider
> >>> networks will not make these concerns and ICE for SIP a moot
> >>> issue, but
> >>> this is another topic.
> >>
> >> Henry,
> >>
> >> I don't quite understand why you are requesting a status change of
> >> ICE? To me it appears that ICE is almost finished spec. It has
> >> been implemented, used and feedback has been influencing the
> >> protocol. It is definitly one of the better examples of current
> >> IETF work that actually has running code. Sure, the
> >> implementations may not be fully updated yet to the latest draft
> >> version. But I think it is safe to say that ICE works in the
> >> intended core cases. If there are some corner cases where it
> >> fails, that may be. But that is not information we will have until
> >> really large scale deployement or someone makes very extensive tests.
> >
> > Henry has, I believe, been very consistent in urging the IETF to
> > have more "running code" experience on every specification (not
> > just ICE) than our process requires. He has, IIRC, also advocated
> > changing the process to require running code before publishing an
> > RFC. He's even suggested at times that one should have running code
> > before submitting a protocol draft.
> >
> > Another point to consider -- I've known Henry to bark at a tree
> > with no squirrel in it occasionally, but most of the time he's
> > pretty reliable, more so than most technology predictors. If he's
> > worried about ICE, it's probably because he's been talking to
> > people who are trying to implement it and they're telling him it is
> > difficult or that they're having performance or reliability issues.
> > I imagine that there are plenty of internal development teams in
> > various companies that are working on this and not necessarily
> > bringing their results back to the IETF. After all, if everybody
> > sees the emperor as fully clothed, they'd be embarrassed to
> > proclaim otherwise. Of course, I wish all of these developers were
> > tied into the IETF, but then I'm sure I don't want to see every ice-
> > implementor's question on the SIP list. That is why we have a sip-
> > implementors mailing list, right?
> >
> > Personally, I think ICE is the best solution we have. We KNOW of
> > cases where hole-punching doesn't work, and I think ICE will
> > produce usable results in most of those cases. But that's just my
> > personal opinion from reading the specs. I haven't implemented ICE,
> > and I haven't used anybody else's implementation. Sure, it's
> > complex, but that's because the underlying problem is complex --
> > there are just too many different behaviors out there in "NAT and
> > firewall land". Of course, I could be wrong, too. It's happened
> > before, once or twice ;-). Just how sure are we that we're right
> > about this?
> >
> > Are we designing a "spruce goose" or a practical system? Only
> > experience will tell us. But some good flight reports from the
> > prototypes could go a long way towards reassuring us of the
> > design's utility.
> >
> > So far we've had a couple of people report hearsay evidence -- that
> > somebody else has an ICE implementation and it works. Any of you
> > folks have personal experience you can share? Have there been
> > prototypes exercised at the interop events?
> >
> > --
> > Dean
> >
> >
> >
> >
> > _______________________________________________
> > Behave mailing list
> > Behave@ietf.org
> > https://www1.ietf.org/mailman/listinfo/behave
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave



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From: Aki Niemi <aki.niemi@nokia.com>
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Subject: Re: [Sip] Comments on draft-ietf-sip-subnot-etags-00
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Hi Dale,

Thanks for the review.

ext sip-bounces@ietf.org wrote:
> Technical issues:
> 
> What if both Suppress-Body-If-Match and SuppressN-Notify-If-Match are
> specified?  We probably don't really care, but the document should
> make it clear if the combination is standardized or not.

The combination makes no sense, and should receive a 400 Bad Request.
Perhaps the following patch to Section 4.2., 1st paragraph:

OLD:

> When creating a conditional SUBSCRIBE request, the subscriber MUST
>    include a conditional header field including an entity-tag to the

NEW:

> When creating a conditional SUBSCRIBE request, the subscriber MUST
>    include a single conditional header field including an entity-tag to the
               ^^^^^^

(I cut out the next technical issue into a separate email with a new
subject, as I think it may require a bit more discussion.)

<snip />

> Nits:
> 
> Section 2.2
> 
>    o  Relatively low rate of actual notifications triggered by actual
>       state changes
> 
> Better phrased "Relatively low rate of notifications triggered by
> state changes".

Fixed.

> Section 3:
> 
> The entity-tag header is "SIP-ETag", but the "SIP-" portion is
> redundant.  I suppose it's too late to change it to just "ETag".

Right, we have SIP-ETag defined in RFC 3903 already.

> Sections 3 and 4.2:
> 
> "two types of Xs" is better phrased "two Xs".

Fixed.

> Section 4.5:
> 
>    2.  If the condition evaluates to true, the notifier sends a 204 (No
>        Notification) response sends no NOTIFY request.
> 
> Should be "sends a 204 (No Notification) response and sends no NOTIFY
> request.

Fixed.

> Section 5.1:
> 
>    The notifier is free to decide the means for generating an
>    entity-tag, except for the special "*" value.
> 
> Better phrased "free to decide how to generate the entity-tag values,
> except that no entity-tag may be the special "*" value."

Much better. Fixed.

> Section 5.3:
> 
>    the notifier MUST suppress the resulting NOTIFY request, and
>    generate a 204 (No Notification) "No Notification" response.
> 
> Should be "a 204 (No Notification) response".

Fixed.

> Sections 8.3 and 8.4:
> 
>    This document registers two new SIP header field names.  These
>    headers are defined by the following information, which has been
>    added to the header fields sub-registry under
>    http://www.iana.org/assignments/sip-parameters.
> 
> This paragraph is duplicated, and it appears to be addressing both
> header registrations together, rather than either one alone.

Changed each instance to:

   This document registers a new SIP header field called
   Suppress-{Body, Notify}-If-Match. This header field is defined by
   the following information, which has been added to the header
   fields sub-registry under
   http://www.iana.org/assignments/sip-parameters.

Cheers,
Aki


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From sip-bounces@ietf.org Thu Jul 19 06:44:44 2007
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Subject: Model for entity-tags in subnot-etags (was: Re: [Sip] Comments on
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I split out this issue from Dale's email into a separate message.

ext sip-bounces@ietf.org wrote:
> The entity-tag values for a single resource are the same across
> subscriptions, but it is not clear whether they are in some sense the
> same between subscriptions for different resources, or for different
> resource-IDs that refer to "the same" underlying resource.  This may
> have significance for renewed subscriptions whose initial requests are
> sent to the same resource AOR but which are ultimately routed to
> different but "equivalent" resource-IDs.

Qian also had similar comments; obviously this seems to be something the
draft does not define clearly enough. I suspect Section 5.1. would need
additional text explaining the model behind etags in more detail.

However, before we try to come up with that text, let's see if we are on
common ground on some basic assumptions (as I suspect we might not be
there 100%). For the most parts the model should be equivalent to that
in HTTP (or at least to my reading of RFC2616).

Here are the assumptions under which subnot-etags are currently defined:

- There is a single authoritative notifier agent responsible for any
single resource.

- A resource is identified using a SIP URI.

- A subscriber subscribes to a resource using the SIP URI.

- With a successful subscription, a subscriber gets access to an entity
from the resource. (Entity is HTTP lingo, and basically means a certain
"view" of the resource state.)

- An entity can be influenced by different external data, e.g., filter
rules, authorization rules, etc.

- There is a one-to-many relationship between the resource and the
entities.

- Each entity has an assigned entity-tag.

- There is a one-to-many relationship between an entity and its versions
(different etag values, of which only one is current).

- An entity-tag must be unique across all versions of all entities from
a resource.

- The rest are implementation details.

Comments?

Cheers,
Aki


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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Francois Audet" <audet@nortel.com>,
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Hi,
=20
Maybe sending the RPH to the UAC is something that the 199 mechanism =
could be used for.
=20
Regards,
=20
Christer
=20
=20
=20


________________________________

	From: Francois Audet [mailto:audet@nortel.com]=20
	Sent: 18. hein=E4kuuta 2007 21:59
	To: Janet P Gunn
	Cc: sip@ietf.org; James M. Polk
	Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
=09
=09
	I will be happy if the document is revised to explain that you cannot =
expect that responses received by a UAC or Proxy will necessarily =
include RPM, in particular in systems where forking occurs, and that =
therefore, applications that use this draft must take it into account.


________________________________

		From: Janet P Gunn [mailto:jgunn6@csc.com]=20
		Sent: Wednesday, July 18, 2007 10:41
		To: Audet, Francois (SC100:3055)
		Cc: sip@ietf.org; James M. Polk
		Subject: RE: [Sip] New draft modifying RFC 4412 for Responses
	=09
	=09

	=09
		In line.=20
	=09
		"Francois Audet" <audet@nortel.com> wrote on 07/18/2007 11:26:54 AM:
	=09
		> > > What I mean is that the error response may not be delivered=20
		> > to the UAC=20
		> > > because a forking proxy will only deliver one final=20
		> > response. If there=20
		> > > are multiple error responses on the individual forks, one being =
the=20
		> > > one with the RPH, and the other being some other response,=20
		> > it is up to=20
		> > > the proxy to decide which one to forward back.
		> > >=20
		> > > So the UAC can not rely on receiving the RPH to "re-intiate" the =

		> > > session or provide some alternative treatment. This may or=20
		> > may not be=20
		> > > a problem depending on the application (that's why I said "if it =

		> > > absolutely has").
		> > >=20
		> >=20
		> > I don't think that will be a problem.=20
		> >=20
		> > For one thing (for the namespaces I am working with) the=20
		> > expectation would be that ALL the responses (except 100 and=20
		> > 403) would have the RPH on the response, so it wouldn't=20
		> > matter which one made it back to the UAC.=20
		>=20
		> I am not talking about multiple provisional responses from the same
		> endpoint.
		>=20
		> I am talking about multiple responses (provisional and final) from=20
		> multiple forks. For example, multiple 4XX responses.=20
	=09
		Yes, I understood that.
		>=20
		> While it is possible that a single forking proxy may decide to=20
		> include the RPH in all responses, that proxy has absolutely no=20
		> control over downstream and upstream proxies who may not include it
		> (they may be proxies in different domains).
		>=20
		> That's the point I'm trying to make. You can not enforce the=20
		> presence RPH in responses in a global system.=20
	=09
		Agree. For the context I am working in, a "global system" is not of =
concern.  I would not expect universal "RPH in responses" in a global =
system.=20
	=09
		>=20
		> > Secondly, the expectation is that the devices responsible for=20
		> > "re-initiating" WOULD be stateful, and not dependant on the=20
		> > information in the RPH in the response.  It is the=20
		> > intervening devices that might not be stateful, and would=20
		> > take advantage of the RPH information in the responses.=20
		>=20
		> That is precisely the type of clarification that needs to be =
included
		> in the draft. It points out the constraints of the mechanism so=20
		> implementors use the feature properly.
		>=20
		> Some text explaining that the UAC needs to be able to deal with the =
RPH
		> not being present in responses gracefully (and why) is needed.=20
	=09
		I am not sure why that needs to be in the ID.  It is an implementation =
specific issue.  =20
	=09
		Since  RFC 4412, as currently published, forbids RPH in responses, I =
think it "goes without saying" that "the UAC needs to be able to deal =
with the RPH not being present in responses".=20
	=09
		Further, while I know that the systems (and namespaces) _I_ work with =
intend to use RPH in responses, I also have been told that other systems =
(using other namespaces) do not intend to put RPH in ANY responses.  I =
do not think there is any need to standardize this aspect, just to =
adjust the standard so it is PERMITTED.
		>=20
		> Similar text for proxies also need to be there. What if a proxy=20
		> forwards a request with RPH and it receives nultiple responses from
		> the downstream proxies, with some including the RPH and others not
		> including it? Does this need to be standardized?=20
	=09
		Again, I do not think there is a need to standardize this aspect.=20
	=09
		Janet
		>=20
		>=20
		> _______________________________________________
		> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
		> This list is for NEW development of the core SIP Protocol
		> Use sip-implementors@cs.columbia.edu for questions on current sip
		> Use sipping@ietf.org for new developments on the application of sip
	=09


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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 6.00.2800.1595" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007>Maybe&nbsp;sending the RPH to the UAC&nbsp;is =
something=20
that&nbsp;the 199 mechanism&nbsp;could be used for.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D621201011-19072007>Christer</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=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> Francois Audet =
[mailto:audet@nortel.com]=20
  <BR><B>Sent:</B> 18. hein=E4kuuta 2007 21:59<BR><B>To:</B> Janet P=20
  Gunn<BR><B>Cc:</B> sip@ietf.org; James M. Polk<BR><B>Subject:</B> RE: =
[Sip]=20
  New draft modifying RFC 4412 for Responses<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN class=3D416025718-18072007><FONT =
face=3DArial=20
  color=3D#800000 size=3D2>I will be happy if the document is revised to =
explain=20
  that you cannot expect that responses received by a UAC or =
Proxy&nbsp;will=20
  necessarily include RPM, in particular in systems where forking =
occurs, and=20
  that therefore, applications that use this draft must take it into=20
  account.</FONT></SPAN></DIV><BR>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 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> Janet P Gunn =
[mailto:jgunn6@csc.com]=20
    <BR><B>Sent:</B> Wednesday, July 18, 2007 10:41<BR><B>To:</B> Audet, =

    Francois (SC100:3055)<BR><B>Cc:</B> sip@ietf.org; James M.=20
    Polk<BR><B>Subject:</B> RE: [Sip] New draft modifying RFC 4412 for=20
    Responses<BR></FONT><BR></DIV>
    <DIV></DIV><BR><FONT face=3Dsans-serif size=3D2><BR>In line.</FONT>=20
    <BR><BR><FONT size=3D2><TT>"Francois Audet" &lt;audet@nortel.com&gt; =
wrote on=20
    07/18/2007 11:26:54 AM:<BR><BR>&gt; &gt; &gt; What I mean is that =
the error=20
    response may not be delivered <BR>&gt; &gt; to the UAC <BR>&gt; &gt; =
&gt;=20
    because a forking proxy will only deliver one final <BR>&gt; &gt; =
response.=20
    If there <BR>&gt; &gt; &gt; are multiple error responses on the =
individual=20
    forks, one being the <BR>&gt; &gt; &gt; one with the RPH, and the =
other=20
    being some other response, <BR>&gt; &gt; it is up to <BR>&gt; &gt; =
&gt; the=20
    proxy to decide which one to forward back.<BR>&gt; &gt; &gt; =
<BR>&gt; &gt;=20
    &gt; So the UAC can not rely on receiving the RPH to "re-intiate" =
the=20
    <BR>&gt; &gt; &gt; session or provide some alternative treatment. =
This may=20
    or <BR>&gt; &gt; may not be <BR>&gt; &gt; &gt; a problem depending =
on the=20
    application (that's why I said "if it <BR>&gt; &gt; &gt; absolutely=20
    has").<BR>&gt; &gt; &gt; <BR>&gt; &gt; <BR>&gt; &gt; I don't think =
that will=20
    be a problem. <BR>&gt; &gt; <BR>&gt; &gt; For one thing (for the =
namespaces=20
    I am working with) the <BR>&gt; &gt; expectation would be that ALL =
the=20
    responses (except 100 and <BR>&gt; &gt; 403) would have the RPH on =
the=20
    response, so it wouldn't <BR>&gt; &gt; matter which one made it back =
to the=20
    UAC. <BR>&gt; <BR>&gt; I am not talking about multiple provisional =
responses=20
    from the same<BR>&gt; endpoint.<BR>&gt; <BR>&gt; I am talking about =
multiple=20
    responses (provisional and final) from <BR>&gt; multiple forks. For =
example,=20
    multiple 4XX responses.</TT></FONT> <BR><BR><FONT size=3D2><TT>Yes, =
I=20
    understood that.<BR>&gt; <BR>&gt; While it is possible that a single =
forking=20
    proxy may decide to <BR>&gt; include the RPH in all responses, that =
proxy=20
    has absolutely no <BR>&gt; control over downstream and upstream =
proxies who=20
    may not include it<BR>&gt; (they may be proxies in different=20
    domains).<BR>&gt; <BR>&gt; That's the point I'm trying to make. You =
can not=20
    enforce the <BR>&gt; presence RPH in responses in a global=20
    system.</TT></FONT> <BR><BR><FONT size=3D2><TT>Agree. For the =
context I am=20
    working in, a "global system" is not of concern. &nbsp;I would not =
expect=20
    universal "RPH in responses" in a global system.</TT></FONT> =
<BR><FONT=20
    size=3D2><TT><BR>&gt; <BR>&gt; &gt; Secondly, the expectation is =
that the=20
    devices responsible for <BR>&gt; &gt; "re-initiating" WOULD be =
stateful, and=20
    not dependant on the <BR>&gt; &gt; information in the RPH in the =
response.=20
    &nbsp;It is the <BR>&gt; &gt; intervening devices that might not be=20
    stateful, and would <BR>&gt; &gt; take advantage of the RPH =
information in=20
    the responses. <BR>&gt; <BR>&gt; That is precisely the type of =
clarification=20
    that needs to be included<BR>&gt; in the draft. It points out the=20
    constraints of the mechanism so <BR>&gt; implementors use the =
feature=20
    properly.<BR>&gt; <BR>&gt; Some text explaining that the UAC needs =
to be=20
    able to deal with the RPH<BR>&gt; not being present in responses =
gracefully=20
    (and why) is needed.</TT></FONT> <BR><BR><FONT size=3D2><TT>I am not =
sure why=20
    that needs to be in the ID. &nbsp;It is an implementation specific =
issue.=20
    &nbsp;</TT></FONT> <BR><BR><FONT size=3D2><TT>Since &nbsp;RFC 4412, =
as=20
    currently published, forbids RPH in responses, I think it "goes =
without=20
    saying" that "the UAC needs to be able to deal with the RPH not =
being=20
    present in responses".</TT></FONT> <BR><BR><FONT =
size=3D2><TT>Further, while I=20
    know that the systems (and namespaces) _I_ work with intend to use =
RPH in=20
    responses, I also have been told that other systems (using other =
namespaces)=20
    do not intend to put RPH in ANY responses. &nbsp;I do not think =
there is any=20
    need to standardize this aspect, just to adjust the standard so it =
is=20
    PERMITTED.<BR>&gt; <BR>&gt; Similar text for proxies also need to be =
there.=20
    What if a proxy <BR>&gt; forwards a request with RPH and it receives =

    nultiple responses from<BR>&gt; the downstream proxies, with some =
including=20
    the RPH and others not<BR>&gt; including it? Does this need to be=20
    standardized?</TT></FONT> <BR><BR><FONT size=3D2><TT>Again, I do not =
think=20
    there is a need to standardize this aspect.</TT></FONT> =
<BR><BR><FONT=20
    size=3D2><TT>Janet<BR>&gt; <BR>&gt; <BR>&gt;=20
    _______________________________________________<BR>&gt; Sip mailing =
list=20
    &nbsp;https://www1.ietf.org/mailman/listinfo/sip<BR>&gt; This list =
is for=20
    NEW development of the core SIP Protocol<BR>&gt; Use=20
    sip-implementors@cs.columbia.edu for questions on current =
sip<BR>&gt; Use=20
    sipping@ietf.org for new developments on the application of=20
  sip<BR></BLOCKQUOTE></BLOCKQUOTE></TT></FONT></BODY></HTML>

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Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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On 18-Jul-07, at 22:56 , Frank W. Miller wrote:

<Algorithm details deleted>
>
> Its just about as simple as it gets and I know you're going to say  
> that
> it doesn't work 100% of the time.  I haven't measured how well its  
> done
> in deployments I've done as yet.  I guess I need to do something along
> those lines.  The good news is, I haven't had anyone complain about it
> so I guess I've been lucky.  So let me have it...  ;)

This is the original STUN algorithm.
One situation where it fails is the following:

                STUN
               server
                 |
                NAT
                 |
         ------------------
         |                |
        UA1              UA2

If the NAT does not support hairpinning, then your algorithm will not
work.

[http://tools.ietf.org/html/draft-jennings-behave-test-results-04
shows that most NATs today do not support UDP hairpinning, and
http://www.guha.cc/saikat/pub/imc05-tcpnat.pdf reports that most do not
support TCP hairpinning.]

This situation can occur when the two UAs belong to the same company,
or both are in the same hotel, or both use the same service provider
and the service provider has a NAT in front of its entire network, etc.


ICE does to things to solve this problem. First of all, UA1 and UA2  
exchange
their local addresses and ports as well as their STUN-learned addresses.
They then test to see which path works. Second, they also exchange
TURN addresses, which serve as a backup in case everything else fails.

David Barrett's algorithm, if I understand it correctly, would work in
this situation. (As far as I can tell, David's is a simplified version
of ICE).

- Philip



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=20

=20

Hi All,

            Can any one please explain, what error code should the
server respond with, when it gets a REGISTER with expired credentials.=20

Is there any standard explaining this behavior.

=20

Thanks,

Anand Kumaravel



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<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'>Hi All,<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'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;
Can any one please explain, what error code should the server respond with,
when it gets a REGISTER with expired credentials.=
 <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'>Is there any standard explaining this=
 behavior.<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'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><b><font size=3D2 color=3D"#3333cc" face=3DArial><span
style=
=3D'font-size:10.0pt;font-family:Arial;color:#3333CC;font-weight:bold'>Anan=
d
Kumaravel<o:p></o:p></span></font></b></p>

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Subject: RE: [Sip] Error code when password expired.
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one option is server can send a 401 Unauthorized with a new set of
credentials. Then the user can re register
=20

Regards=20
Ranjit=20

=20

________________________________

From: Anand Kumaravel, TLS-Chennai [mailto:akumaravel@hcl.in]=20
Sent: Thursday, July 19, 2007 4:00 PM
To: sip@ietf.org
Subject: [Sip] Error code when password expired.



=20

=20

Hi All,

            Can any one please explain, what error code should the
server respond with, when it gets a REGISTER with expired credentials.=20

Is there any standard explaining this behavior.

=20

Thanks,

Anand Kumaravel

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<DIV dir=3Dltr align=3Dleft><SPAN class=3D769392312-19072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>one option is server can send a 401 =
Unauthorized with a new=20
set of credentials. Then the user can re register</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3D"Lucida Sans" color=3D#0000ff=20
size=3D2>Regards</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
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color=3D#0000ff size=3D2>Ranjit</FONT></SPAN> </P>
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<FONT face=3DTahoma size=3D2><B>From:</B> Anand Kumaravel, TLS-Chennai=20
[mailto:akumaravel@hcl.in] <BR><B>Sent:</B> Thursday, July 19, 2007 4:00 =

PM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Error code when =
password=20
expired.<BR></FONT><BR></DIV>
<DIV></DIV>
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<P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
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<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
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<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hi =
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<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
Can any one please explain, what error code should the server respond =
with, when=20
it gets a REGISTER with expired credentials. =
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is there any standard =
explaining=20
this behavior.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
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<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Thanks,<o:p></o:p></SPAN></FONT></P>
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size=3D2><SPAN=20
style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #3333cc; =
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Kumaravel<o:p></o:p></SPAN></FONT></B></P></DIV>
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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
--===============0385345172==--





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Juha Heinanen wrote:
> a while back i read ice tutorial slides and my impression was that the
> protocol was very complex (pity implementers) and provided only a bit
> better nat traversal capability than what [open]ser/rtpproxy/mediaproxy
> have been offering to lots of users for years already.

 The NAT traversal algorithm implemented in [open]ser, rtpproxy, and
 mediaproxy had been developed long before the first version of the ICE
 draft; at that time no other solution existed (except for preliminary
 STUN and TURN drafts), and we needed something to make it work; in
 addition to that NAT traversal support in UA was virtually non-existent
 back then.

 Having said that, it is true that the algorithm implemented in
 [open]ser is simpler than ICE, but I believe that these two cannot
 be directly compared. Even the NAT traversal logic in SER can get
 pretty complex if none of the end-points is a PSTN gateway in the
 public internet (especially if you want to handle re-INVITEs and
 serial forking properly).

   Jan.


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The stale parameter can also be indicated within the 401 response; see
rfc2617 section 3.3.
=20


________________________________

	From: Avasarala Ranjit-A20990 [mailto:ranjit@motorola.com]=20
	Sent: Thursday, July 19, 2007 8:24 AM
	To: Anand Kumaravel, TLS-Chennai; sip@ietf.org
	Subject: RE: [Sip] Error code when password expired.
=09
=09
	one option is server can send a 401 Unauthorized with a new set
of credentials. Then the user can re register
	=20

	Regards=20
	Ranjit=20

	=20

________________________________

	From: Anand Kumaravel, TLS-Chennai [mailto:akumaravel@hcl.in]=20
	Sent: Thursday, July 19, 2007 4:00 PM
	To: sip@ietf.org
	Subject: [Sip] Error code when password expired.
=09
=09

	=20

	=20

	Hi All,

	            Can any one please explain, what error code should
the server respond with, when it gets a REGISTER with expired
credentials.=20

	Is there any standard explaining this behavior.

	=20

	Thanks,

	Anand Kumaravel

DISCLAIMER:
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The contents of this e-mail and any attachment(s) are confidential and
intended for the named recipient(s) only.
It shall not attach any liability on the originator or HCL or its
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reflect the opinions of HCL or its affiliates.
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modification, distribution and / or publication of=20
this message without the prior written consent of the author of this
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class=3D625533212-19072007>The stale parameter can also be indicated =
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401 response; see rfc2617 section 3.3.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
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  <DIV dir=3Dltr align=3Dleft><SPAN class=3D769392312-19072007><FONT =
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  color=3D#0000ff size=3D2>one option is server can send a 401 =
Unauthorized with a=20
  new set of credentials. Then the user can re =
register</FONT></SPAN></DIV>
  <DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
  <P><SPAN lang=3Den-us><FONT face=3D"Lucida Sans" color=3D#0000ff=20
  size=3D2>Regards</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
face=3D"Lucida Sans"=20
  color=3D#0000ff size=3D2>Ranjit</FONT></SPAN> </P>
  <DIV>&nbsp;</DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Anand Kumaravel, TLS-Chennai =

  [mailto:akumaravel@hcl.in] <BR><B>Sent:</B> Thursday, July 19, 2007 =
4:00=20
  PM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Error code when =

  password expired.<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hi=20
  All,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
=20
  Can any one please explain, what error code should the server respond =
with,=20
  when it gets a REGISTER with expired credentials.=20
<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is there any standard =
explaining=20
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  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
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  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Thanks,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><B><FONT face=3DArial color=3D#3333cc =
size=3D2><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: #3333cc; =
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  Kumaravel<o:p></o:p></SPAN></FONT></B></P></DIV>
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Gonzalo Camarillo wrote:
> Hi Paul,
> 
>> Thanks for sending out in advance. Maybe we can get into the 
>> discussion of the slides now, before the meeting.
> 
> yes, that was the intention.
> 
>> Regarding references to body parts:
>>
>> IMO it is a *bad* idea to make the decision about processing of a body 
>> part "on its own" dependent on whether the node doing so has detected 
>> a reference to that body part or not.
>>
>> For one thing, that requires a multi-pass parsing mechanism to check 
>> for references before deciding what processing to do on a particular 
>> part. For another, it requires that the recipient understand every 
>> reference. For instance, the sender may have included an extension 
>> header that contains a reference to a body part. If the extension 
>> isn't understood, then the body part should not be processed. But if 
>> the recipient doesn't understand the extension then it won't recognize 
>> the reference either, and so will attempt to process the body part as 
>> one without references.
> 
> yes, these two issues are described in the draft.
> 
>> The only good solution I see to this is to partition the 
>> Content-Disposition and/or Content-Type values whose processing is 
>> dependent on reference from those whose processing is not dependent on 
>> a reference. For instance, we could simply define a new 
>> Content-Disposition of "by-reference".
>>
>> Then a normal sequence of processing would be to first process all the 
>> body parts according to their type and disposition alone. Any with 
>> disposition "by-reference" would be set aside. Then, when processing 
>> headers, if a header with a reference is encountered the appropriate 
>> processing could be applied to the set aside body part. References 
>> within other body parts would be a little more complex, but you get 
>> the idea.
> 
> yes. When looking for references, one would need to process the header 
> fields plus the body parts with a disposition type different than 
> 'by-reference', since, as you point out, those body parts could contain 
> references to 'by-reference' body parts.
> 
> The only case we would not cover would be two (or more) 'by-reference' 
> body parts referencing each other. But we can just say that this is 
> forbidden (no extension does that at present).

Right. That case makes no sense to construct.

>> This would not prevent reference to body parts with other 
>> dispositions. It just means that those body parts would also be 
>> processed according to their type and disposition, in addition to any 
>> processing they might get according to their reference. The 
>> "by-reference" disposition is just a defintion that gets no implicit 
>> processing.
> 
> yes, the draft already talks about the possibility of having several 
> references to a body part. In such a case, the body part would need to 
> be processed several times (in different ways). However, the case you 
> are talking about is not currently addresses (i.e., a body part is 
> processed once based on a reference and another time based on its 
> content disposition). None of these situations occur within current 
> extensions, but they could happen in the future.

Yes. IMO it is important to get a consistent framework in place that 
covers whatever kinds of usage might come along. Without that people may 
invent processing algorithms that cover only the cases they happen to 
have encountered.

>> An advantage of this approach is that by-reference body parts don't 
>> need to be parsed unless the reference is processed and the action 
>> required in processing the reference requires the node to parse the 
>> referenced body. (Interaction with handling=required in this case is 
>> TBD.) Even a multipart with disposition of by-reference might not need 
>> to be parsed.
> 
> The interaction with handling=required would be that if you are not able 
> to find a reference to a 'by-reference' body part with 
> handling=required, you report an error because you have not been able to 
> process the body part.

Hmm. I guess that makes sense. It means that any time there is an 
optional header with a reference to a body part then the handling of 
that body part needs to be optional. But again I guess that makes good 
sense.

Some interesting questions arise when body parts are being processed by 
middleboxes. (Maybe we aren't supposed to talk about that here.) The 
whole issue of handling=required gets messy there. The middlebox isn't 
the intended recipient, so the handling parameter doesn't really apply 
to it. So presumably it processes whatever headers and body parts it 
likes. That is another motivation for making the decision about 
processing be explicit - so it can be selective.

> This approach would work and has the advantage to be explicit about how 
> a body part is expected to be processed... I would like to hear more 
> opinions on this.

Me too. So far this has mostly been a dialog between the two of us.

(Dale? - I can usually count on you to have an opinion.)

	Paul

> Cheers,
> 
> Gonzalo
> 
> 
>> Gonzalo Camarillo wrote:
>>> Folks,
>>>
>>> here you have the slides I have put together for the body (MIME) 
>>> handling presentation on Monday:
>>>
>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>
>>>
>>> These slides relate to this draft:
>>>
>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>
>>>
>>> The slides discuss all the open issues that have been brought up in 
>>> the list in the last months.
>>>
>>> Comments are welcome.
>>>
>>> Thanks,
>>>
>>> Gonzalo
> 


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Gonzalo Camarillo wrote:
> Hi Paul,
> 
>> Regarding Content-Disposition in multiparts:
>>
>> While it seems "wrong" to me, I can live with using "render" as the 
>> normal disposition for multipart, *if* we can do a careful job of 
>> defining what "render" actually means for SIP. Used this way it 
>> clearly doesn't always mean to display the content to a user.
> 
> An entity receiving a 'multipart/mixed' body will process its body parts 
> based on their disposition types.If the disposition type of a particular 
> body part is not understood, it will be displayed to the user because 
> the default disposition type is 'render'.

There is a bigger problem with 'render' than just multipart.

We have many uses of body parts that never mention Content-Disposition 
at all. The related examples don't use it, and so they all get a default 
of 'render'. This is particularly the case with events. E.g.

- the notification for REFER contains type application/sipfrag.
   Its probably not intended to be "rendered" in any literal sense,
   though some conclusion about success or failure may be.

- usage of filter bodies in presence subscriptions (RFC 4660)
   doesn't have a specified disposition, so it implicitly uses render.
   Its hard for me to convince myself that what the presence server
   does with these can be described as rendering.

- In fact, the only place I can think of where 'render' is (implicitly)
   used in sip and has the appropriate connotation is in MESSAGE.

I think we just have to grandfather the use of 'refer' in SIP, and 
document that the usage in sip does not mean "render", but rather means 
that the usage must be determined by other context - specifically the 
type of message or response and the content type. IOW, 'render' in sip 
can be thought of as a peculiar spelling for 'by-context'. It might be 
helpful to give guidance for development of future specs about when 
'render' should be used vs defining a new disposition.

> Therefore, the 'multipart/mixed' body will never be processed based on 
> its disposition type.

Based on my discussion above, I would consider this a case of the 
multipart being processed according to its context *because* its 
content-disposition is 'refer'. If it had some other content-disposition 
then that might mean that it should be processed specially. E.g. if a 
multipart/mixed had disposition 'by-reference' then that needs to be 
taken into account, and the processing of the multipart becomes 
conditional on the reference.

Also, there are only certain contexts in which multipart should be 
processed "normally". Consider a NOTIFY from a REFER, with a 
Content-Type of message/sipfrag, which happened to include a body as 
part of the fragment. (This is legal.) That body might contain a 
multipart/mixed, and that might contain other stuff. But I think the 
processing of all of that is dependent on processing of the sipfrag. For 
instance, if something inside that had handling=required, I don't think 
that means that processing of the NOTIFY fails because the recipient 
can't handle that.

> This means that it does not make much sense to 
> talk about its disposition type as a whole.

Counterexample above.

> However, we need to have one 
> in order to be able to set its handling parameter. That's why we can 
> either keep 'render' explaining that it is actually never used or define 
> a new disposition type.

I can go either way on that. But because there is existing usage, and 
because 'render' semantics are already so messed up, I think it makes 
more sense to stick with 'render' and just clarify its meaning.

> Since existing implementations will not understand our newly defined 
> disposition type, they will treat it as 'render' anyway... but the 
> resulting behavior would be the same. That's why it may make sense to 
> keep using 'render'.

Yup.

> On the other hand, semantically, it would make more sense to define a 
> new disposition type.
> 
> Also, if we decide to define the 'by-reference' disposition type (per a 
> different email thread), it could be used here as well.

It could be used when there is actually a reference to it. But it would 
not be an appropriate disposition to use when there is no reference to 
the multipart.

	Thanks,
	Paul

> Cheers,
> 
> Gonzalo
> 
> 
> 
>> Gonzalo Camarillo wrote:
>>> Folks,
>>>
>>> here you have the slides I have put together for the body (MIME) 
>>> handling presentation on Monday:
>>>
>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>
>>>
>>> These slides relate to this draft:
>>>
>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>
>>>
>>> The slides discuss all the open issues that have been brought up in 
>>> the list in the last months.
>>>
>>> Comments are welcome.
>>>
>>> Thanks,
>>>
>>> Gonzalo
>>>
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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There are in fact deployments of ICE, more than one, based on previous 
versions of the draft.

Most folks I talk to are waiting to IETF to finish this thing, so they 
can go upgrade their code and move to the final version. Delaying it, 
making it experiental, informational, or any other variation will serve 
no other purpose than to convince everyone that this work is not done, 
and so folks should continue to wait. If you want an industry's worth of 
feedback and data, lets go and finish this thing. That means PS.

All of that said, ICE has, probably more than most other specs that have 
ever come out of RAI, have had the benefit of:

   a. extensive technical review, comments and feedbacks, including 
commentary from implementors (including one in my own company, and 
several others from outside of it) and deployers (Tim Moore from Microsoft)

   b. design based on years of experience with NATs and NAT traversal

   c. design based on real collected data on NAT behaviors and on 
techniques used in other applications

   d. consideration on incremental deployability and applicability 
across the entire range of SIP, from p2p to enterprise to large provider 
to closed application provider and so on

   e. multiple open source implementations (two have been pointed out 
already)

   f. an active design team including security experts that helped in 
the massive redesign in -10 to simplify things

And as such, it would seem to me that, not only is there not a single 
shred of evidence that suggests this draft deserves anything less than 
PS, there is evidence only to demonstrate that it has quality far 
greater than most IETF protocols with that moniker.

Other parts of this thread have been folks suggesting alternative 
techniques that are simpler or work better. Well, I look forward to 
their drafts. So far, no one has suggested a single thing which was not 
considered and/or incorporated into ICE in some way over the last four 
years.

ICE was built for reliability - to cover lots of cases and achieve not 
just 90% effectiveness (not nearly good enough), but as effective as is 
possible within the limitations of hurting the Internet or violating 
security policies and other things we won't touch. Without a doubt you 
can design a custom protocol for any one single deployment scenario with 
some set of assumptions that works simpler with fewer messages. However, 
our objective was to design a NAT traversal solution for SIP that 
applied as broadly as SIP did. ICE has the benefit of resulting in 
simple flows in simple topologies, but the same logic produces more 
complex flows in more complex situations, but still works. That was a 
design decision we made years ago and it remains a sound one.

Comments have also been made that, "well, you thought STUN would work 
and you were wrong". This too is bogus. The limitations of STUN were 
known at the time of its publication. It was our hope that enough of a 
market would find it acceptable to be useful. Quoting from RFC 3489:

>  STUN introduces brittleness into the system in several ways:
> 
>    o  The discovery process assumes a certain classification of devices
>       based on their treatment of UDP.  There could be other types of
>       NATs that are deployed that would not fit into one of these molds.
>       Therefore, future NATs may not be properly detected by STUN.  STUN
>       clients (but not servers) would need to change to accommodate
>       that.
> 
>    o  The binding acquisition usage of STUN does not work for all NAT
>       types.  It will work for any application for full cone NATs only.
>       For restricted cone and port restricted cone NAT, it will work for
>       some applications depending on the application. Application
>       specific processing will generally be needed.  For symmetric NATs,
>       the binding acquisition will not yield a usable address.  The
>       tight dependency on the specific type of NAT makes the protocol
>       brittle.
> 
>    o  STUN assumes that the server exists on the public Internet.  If
>       the server is located in another private address realm, the user
>       may or may not be able to use its discovered address to
>       communicate with other users.  There is no way to detect such a
>       condition.
> 
>    o  The bindings allocated from the NAT need to be continuously
>       refreshed.  Since the timeouts for these bindings is very
>       implementation specific, the refresh interval cannot easily be
>       determined.  When the binding is not being actively used to
>       receive traffic, but to wait for an incoming message, the binding
>       refresh will needlessly consume network bandwidth.
> 
>    o  The use of the STUN server as an additional network element
>       introduces another point of potential security attack.  These
>       attacks are largely prevented by the security measures provided by
>       STUN, but not entirely.
> 
> 
> 
> 
> 
> 
> 
> Rosenberg, et al.           Standards Track                    [Page 40]
> 
> RFC 3489                          STUN                        March 2003
> 
> 
>    o  The use of the STUN server as an additional network element
>       introduces another point of failure.  If the client cannot locate
>       a STUN server, or if the server should be unavailable due to
>       failure, the application cannot function.
> 
>    o  The use of STUN to discover address bindings will result in an
>       increase in latency for applications.  For example, a Voice over
>       IP application will see an increase of call setup delays equal to
>       at least one RTT to the STUN server.
> 
>    o  The discovery of binding lifetimes is prone to error.  It assumes
>       that the same lifetime will exist for all bindings. This may not
>       be true if the NAT uses dynamic binding lifetimes to handle
>       overload, or if the NAT itself reboots during the discovery
>       process.
> 
>    o  STUN imposes some restrictions on the network topologies for
>       proper operation.  If client A obtains an address from STUN server
>       X, and sends it to client B, B may not be able to send to A using
>       that IP address.  The address will not work if any of the
>       following is true:
> 
>       -  The STUN server is not in an address realm that is a common
>          ancestor (topologically) of both clients A and B.  For example,
>          consider client A and B, both of which have residential NAT
>          devices.  Both devices connect them to their cable operators,
>          but both clients have different providers. Each provider has a
>          NAT in front of their entire network, connecting it to the
>          public Internet.  If the STUN server used by A is in A's cable
>          operator's network, an address obtained by it will not be
>          usable by B.  The STUN server must be in the network which is a
>          common ancestor to both - in this case, the public Internet.
> 
>       -  The STUN server is in an address realm that is a common
>          ancestor to both clients, but both clients are behind the same
>          NAT connecting to that address realm.  For example, if the two
>          clients in the previous example had the same cable operator,
>          that cable operator had a single NAT connecting their network
>          to the public Internet, and the STUN server was on the public
>          Internet, the address obtained by A would not be usable by B.
>          That is because some NATs will not accept an internal packet
>          sent to a public IP address which is mapped back to an internal
>          address.  To deal with this, additional protocol mechanisms or
>          configuration parameters need to be introduced which detect
>          this case.
> 
> 
> 
> 
> 
> 
> Rosenberg, et al.           Standards Track                    [Page 41]
> 
> RFC 3489                          STUN                        March 2003
> 
> 
>    o  Most significantly, STUN introduces potential security threats
>       which cannot be eliminated.  This specification describes
>       heuristics that can be used to mitigate the problem, but it is
>       provably unsolvable given what STUN is trying to accomplish.
>       These security problems are described fully in Section 12.


Well, the market didn't find it useful enough. And so we worked to 
improve. Nearly every one of these limitations has been eliminated by 
ICE. Go read the IAB considerations section, which discusses this. 
Binding lifetime discovery is pretty much the only remaining one; 
fortunately that is a minor one and ICE is just aggressive in that area 
to compensate. The STUN-control work Dan has been promoting is meant to 
add ontop of ICE to finally close that issue.

So, in case its not abundantly clear, I think there is no merit to the 
calls for changing ICE to informational or experimental, and that these 
last minute claims of better solutions or simpler solutions are also 
without merit. ICE is as deserving, and indeed more so, of a PS moniker 
than most other documents that achieve that label.

-Jonathan R.


David Barrett wrote:
> Hi Cullen, I think you bring up lot of really great and detailed questions.
> 
> However, speaking for myself at least, answering those questions doesn't
> really address my concern.  All I want to know is:
> 
> 	Does ICE work, and how well?
> 
> What "works" means is entirely open to debate.  I've tossed out "high peer
> connectivity" and "low connection setup time" as the most important
> criteria, and I've put hard data behind those numbers as a stake in the
> ground for comparison, but I'm open to other metrics as well.
> 
> But regardless of the metrics, I want somebody who uses it and loves it and
> bases their business upon its success to stand up and explain why it's
> great.  And I want this person to provide real-world data, sampled on a wide
> scale, proving that ICE "works" -- by any definition.
> 
> Basically, I want ICE firmly grounded in wide-scale testing and real-world
> data, and answering the questions below doesn't bring us any closer to that.
> Whether it's level 10 or level 12 doesn't produce new data.  How its
> algorithms differ from competing ones doesn't show that they're any better
> or worse.  Detailed proposals for change don't shed any light on whether
> those changes improve or worsen it, because its current state is completely
> unmeasured.
> 
> The most powerful thing the IETF can do to standardize the world upon ICE is
> prove that it works better than any alternative.  Because until that's
> proven, we just need to take it as a matter of faith.
> 
> -david
> 
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Wednesday, July 18, 2007 7:56 PM
>> To: SIP; Behave WG; mmusic@ietf.org
>> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>>
>> First I apologize for the cross post - this is a mmusic draft, I
>> would appreciate replies on the mmusic list only.
>>
>> Few things I am looking for here  ....
>>
>> 1) specific reasons why this work is less appropriate for standards
>> track than other work that IETF puts on the standards track. Keep in
>> mind the first step of standards track is Proposed Standards which
>> requires basically no interoperability, then it could move to Draft
>> Standard where interoperability is shown and finally Full Standard if
>> it turns out to be a good idea.  If you are going to post a reply to
>> this point, make sure you have read RFC 2026.
>>
>> 2) I am interested in the specifics of problems that ICE fails to
>> solve that the WG has not previous considered. There are lots of
>> people that have implemented versions of it and think it is a good
>> idea - if people are having some problems with it, I'd love to hear
>> about details of things that are wrong. That it has a lot of messages
>> in some use cases has been known for a long time. It used to have far
>> more messages  - engineering was done to consider what was a
>> reasonable amount for call set up times and a new design was done to
>> bring the message to a level most people invovled with the design
>> felt was reasonable. If there are specific other than these that are
>> wrong and not considered in the design, now would be a good time to
>> get them on the table.
>>
>> 3) If people think that Skype for example has a simpler solution that
>> works better than ICE. Feel free to describe exactly what it is.
>> Several people invovled with the ICE design have looked in extreme
>> detail at Skype and others and so far none of them have come back
>> with what we should do different than ICE.
>>
>> 4) ICE can be used in a mode where it only provided the reflexive
>> candidate. People were pretty clear that this was far inferior to the
>> modes where there are other candidates. If someone would like to
>> provide exactly how hole punching works better than this very limited
>> ICE technique, I'd be glad to hear the details. This has certainly
>> been an topic that any could have brought up over the last several
>> years. If someone has specifics, I suggest they get out a full
>> proposal really soon. I am very doubtful that hole punching solution
>> is fundamentally different than the type of hole punching ICE does.
>>
>> 5) Some people have suggested that if all NATs were behave complaint,
>> then we would not need ICE. I strongly disagree. I think the only
>> part of ICE we would not need is the TURN relays which are more or
>> less an optional part. Again, specifics on what can be removed are
>> welcome.
>>
>> ICE is far more complicated that anyone wants - the problem is, no
>> one has come up with a proposal that looks like it comes anywhere
>> close to meeting the requirements and if simpler. The design space
>> has been extensively explored by many people. The raw data to
>> understand what will work on what NATs have been painfully tested and
>> documented in publications such as:
>>
>> http://nutss.gforge.cis.cornell.edu/pub/imc05-tcpnat.pdf
>>
>> http://tools.ietf.org/id/draft-jennings-behave-test-results-04.txt
>>
>> https://www.guha.cc/saikat/stunt-results.php?
>>
>> (Ford used to have a page too but it seems to be gone)
>>
>> I have a very open mind to considering better ways to do ICE - but I
>> need specific suggestions or it is impossible for me to really think
>> about if the suggestions represent a better path or not.
>>
>> I would love to see more open source ICE code (there is lots already)
>> - I feel that this is an area where ICE is likely just table stakes
>> and owning an proprietary ICE stack will not turn out to be a
>> competitive advantage. Having open source implementations where the
>> development cost was shared and everyone could get working products
>> to market faster would be an advantage to companies. Clearly I expect
>> the bulk of ICE implementations to not be open source but I still
>> would be keen on doing what I can to contribute to an open source
>> version as personally I feel that is in the interest of end users, my
>> employer, and the IETF as a SDO.
>>
>> Cullen <with my individual hat on>
>>
>>
>>
>> On Jul 18, 2007, at 9:12 AM, Dean Willis wrote:
>>
>>> On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:
>>>
>>>> Henry Sinnreich skrev:
>>>>> Following some discussions and soul searching, the best approach for
>>>>> ICE-17 for LC would be to _go ahead and make it an experimental
>>>>> RFC_.
>>>>> This would support the development of interoperable
>>>>> implementations, the
>>>>> collection of deployment data and also hopefully open source code.
>>>>> Though I have full faith in ICE, following the practices that
>>>>> made the
>>>>> IETF successful take precedent here, I believe.
>>>>> This is a change from the attached that I wrote on 7/16/2007.
>>>>> Note: We can only hope the proliferation of SBC in VoIP service
>>>>> provider
>>>>> networks will not make these concerns and ICE for SIP a moot
>>>>> issue, but
>>>>> this is another topic.
>>>> Henry,
>>>>
>>>> I don't quite understand why you are requesting a status change of
>>>> ICE? To me it appears that ICE is almost finished spec. It has
>>>> been implemented, used and feedback has been influencing the
>>>> protocol. It is definitly one of the better examples of current
>>>> IETF work that actually has running code. Sure, the
>>>> implementations may not be fully updated yet to the latest draft
>>>> version. But I think it is safe to say that ICE works in the
>>>> intended core cases. If there are some corner cases where it
>>>> fails, that may be. But that is not information we will have until
>>>> really large scale deployement or someone makes very extensive tests.
>>> Henry has, I believe, been very consistent in urging the IETF to
>>> have more "running code" experience on every specification (not
>>> just ICE) than our process requires. He has, IIRC, also advocated
>>> changing the process to require running code before publishing an
>>> RFC. He's even suggested at times that one should have running code
>>> before submitting a protocol draft.
>>>
>>> Another point to consider -- I've known Henry to bark at a tree
>>> with no squirrel in it occasionally, but most of the time he's
>>> pretty reliable, more so than most technology predictors. If he's
>>> worried about ICE, it's probably because he's been talking to
>>> people who are trying to implement it and they're telling him it is
>>> difficult or that they're having performance or reliability issues.
>>> I imagine that there are plenty of internal development teams in
>>> various companies that are working on this and not necessarily
>>> bringing their results back to the IETF. After all, if everybody
>>> sees the emperor as fully clothed, they'd be embarrassed to
>>> proclaim otherwise. Of course, I wish all of these developers were
>>> tied into the IETF, but then I'm sure I don't want to see every ice-
>>> implementor's question on the SIP list. That is why we have a sip-
>>> implementors mailing list, right?
>>>
>>> Personally, I think ICE is the best solution we have. We KNOW of
>>> cases where hole-punching doesn't work, and I think ICE will
>>> produce usable results in most of those cases. But that's just my
>>> personal opinion from reading the specs. I haven't implemented ICE,
>>> and I haven't used anybody else's implementation. Sure, it's
>>> complex, but that's because the underlying problem is complex --
>>> there are just too many different behaviors out there in "NAT and
>>> firewall land". Of course, I could be wrong, too. It's happened
>>> before, once or twice ;-). Just how sure are we that we're right
>>> about this?
>>>
>>> Are we designing a "spruce goose" or a practical system? Only
>>> experience will tell us. But some good flight reports from the
>>> prototypes could go a long way towards reassuring us of the
>>> design's utility.
>>>
>>> So far we've had a couple of people report hearsay evidence -- that
>>> somebody else has an ICE implementation and it works. Any of you
>>> folks have personal experience you can share? Have there been
>>> prototypes exercised at the interop events?
>>>
>>> --
>>> Dean
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Behave mailing list
>>> Behave@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/behave
>>
>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: Re: Model for entity-tags in subnot-etags (was: Re: [Sip] Comments on
	draft-ietf-sip-subnot-etags-00)
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   From: Aki Niemi <aki.niemi@nokia.com>

   - There is a single authoritative notifier agent responsible for any
   single resource.

   - A resource is identified using a SIP URI.

   - A subscriber subscribes to a resource using the SIP URI.

Often a subscriber will send the SUBSCRIBE to a URI which is not the
request-URI of the SUBSCRIBE when it reaches the notifier agent.  This
happens due to normal SIP request routing, but the routing can also
change with time.

   - Each entity has an assigned entity-tag.

Each entity has a succession of versions.  Each version of each entity
has an assigned entity-tag.

   - There is a one-to-many relationship between an entity and its versions
   (different etag values, of which only one is current).

   - An entity-tag must be unique across all versions of all entities from
   a resource.

Does this rule out hash-based entity-tags, which are only
probabalistically unique?

Dale


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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
Date: Thu, 19 Jul 2007 10:22:40 -0400
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On Jul 19, 2007, at 6:15 AM, David Barrett wrote:

> Hi Cullen, I think you bring up lot of really great and detailed =20
> questions.
>
> However, speaking for myself at least, answering those questions =20
> doesn't
> really address my concern.  All I want to know is:
>
> 	Does ICE work, and how well?
>
> What "works" means is entirely open to debate.
It may have been in the past, but the debate I think is closed, =20
because the requirements captured the definition of "works". If you =20
want to re-open the requirements discussion, please be clear about that.

> I've tossed out "high peer
> connectivity" and "low connection setup time" as the most important
> criteria, and I've put hard data behind those numbers as a stake in =20=

> the
> ground for comparison, but I'm open to other metrics as well.
>
See the requirements. One not often mentioned but clearly a =20
requirement that ICE works very hard to meet is "use the most direct =20
path available".

> But regardless of the metrics, I want somebody who uses it and =20
> loves it and
> bases their business upon its success to stand up and explain why it's
> great.
> And I want this person to provide real-world data, sampled on a wide
> scale, proving that ICE "works" -- by any definition.
>
This is in fact an extremely important exercise, and precisely the =20
activity which according to RFC2026 is what should be happening after =20=

approval as proposed standard, and before advancement to  draft =20
standard.

> Basically, I want ICE firmly grounded in wide-scale testing and =20
> real-world
> data, and answering the questions below doesn't bring us any closer =20=

> to that.
> Whether it's level 10 or level 12 doesn't produce new data.  How its
> algorithms differ from competing ones doesn't show that they're any =20=

> better
> or worse.  Detailed proposals for change don't shed any light on =20
> whether
> those changes improve or worsen it, because its current state is =20
> completely
> unmeasured.
>
Showing how the algorithms work better, or work in more cases that =20
are covered by the requirements does in fact show that it's better, =20
with respect to the requirements. Again, you're arguing indirectly =20
that we should re-open the requirements discussion, which I do not =20
think would be terribly productive, but would support if there is in =20
fact consensus to do so.

> The most powerful thing the IETF can do to standardize the world =20
> upon ICE is
> prove that it works better than any alternative.  Because until that's
> proven, we just need to take it as a matter of faith.
>
I do not think that is desirable. The perfect is the enemy of the =20
good. ICE does not have to be better than anything else. It just has =20
to be "good enough" and see wide adoption.

That's my 2=A2 (as an individual, not as an IAB member, in case anybody =20=

cares).

DaveO
> -david
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Wednesday, July 18, 2007 7:56 PM
>> To: SIP; Behave WG; mmusic@ietf.org
>> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC =20
>> for RFC
>>
>>
>> First I apologize for the cross post - this is a mmusic draft, I
>> would appreciate replies on the mmusic list only.
>>
>> Few things I am looking for here  ....
>>
>> 1) specific reasons why this work is less appropriate for standards
>> track than other work that IETF puts on the standards track. Keep in
>> mind the first step of standards track is Proposed Standards which
>> requires basically no interoperability, then it could move to Draft
>> Standard where interoperability is shown and finally Full Standard if
>> it turns out to be a good idea.  If you are going to post a reply to
>> this point, make sure you have read RFC 2026.
>>
>> 2) I am interested in the specifics of problems that ICE fails to
>> solve that the WG has not previous considered. There are lots of
>> people that have implemented versions of it and think it is a good
>> idea - if people are having some problems with it, I'd love to hear
>> about details of things that are wrong. That it has a lot of messages
>> in some use cases has been known for a long time. It used to have far
>> more messages  - engineering was done to consider what was a
>> reasonable amount for call set up times and a new design was done to
>> bring the message to a level most people invovled with the design
>> felt was reasonable. If there are specific other than these that are
>> wrong and not considered in the design, now would be a good time to
>> get them on the table.
>>
>> 3) If people think that Skype for example has a simpler solution that
>> works better than ICE. Feel free to describe exactly what it is.
>> Several people invovled with the ICE design have looked in extreme
>> detail at Skype and others and so far none of them have come back
>> with what we should do different than ICE.
>>
>> 4) ICE can be used in a mode where it only provided the reflexive
>> candidate. People were pretty clear that this was far inferior to the
>> modes where there are other candidates. If someone would like to
>> provide exactly how hole punching works better than this very limited
>> ICE technique, I'd be glad to hear the details. This has certainly
>> been an topic that any could have brought up over the last several
>> years. If someone has specifics, I suggest they get out a full
>> proposal really soon. I am very doubtful that hole punching solution
>> is fundamentally different than the type of hole punching ICE does.
>>
>> 5) Some people have suggested that if all NATs were behave complaint,
>> then we would not need ICE. I strongly disagree. I think the only
>> part of ICE we would not need is the TURN relays which are more or
>> less an optional part. Again, specifics on what can be removed are
>> welcome.
>>
>> ICE is far more complicated that anyone wants - the problem is, no
>> one has come up with a proposal that looks like it comes anywhere
>> close to meeting the requirements and if simpler. The design space
>> has been extensively explored by many people. The raw data to
>> understand what will work on what NATs have been painfully tested and
>> documented in publications such as:
>>
>> http://nutss.gforge.cis.cornell.edu/pub/imc05-tcpnat.pdf
>>
>> http://tools.ietf.org/id/draft-jennings-behave-test-results-04.txt
>>
>> https://www.guha.cc/saikat/stunt-results.php?
>>
>> (Ford used to have a page too but it seems to be gone)
>>
>> I have a very open mind to considering better ways to do ICE - but I
>> need specific suggestions or it is impossible for me to really think
>> about if the suggestions represent a better path or not.
>>
>> I would love to see more open source ICE code (there is lots already)
>> - I feel that this is an area where ICE is likely just table stakes
>> and owning an proprietary ICE stack will not turn out to be a
>> competitive advantage. Having open source implementations where the
>> development cost was shared and everyone could get working products
>> to market faster would be an advantage to companies. Clearly I expect
>> the bulk of ICE implementations to not be open source but I still
>> would be keen on doing what I can to contribute to an open source
>> version as personally I feel that is in the interest of end users, my
>> employer, and the IETF as a SDO.
>>
>> Cullen <with my individual hat on>
>>
>>
>>
>> On Jul 18, 2007, at 9:12 AM, Dean Willis wrote:
>>
>>>
>>> On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:
>>>
>>>> Henry Sinnreich skrev:
>>>>> Following some discussions and soul searching, the best =20
>>>>> approach for
>>>>> ICE-17 for LC would be to _go ahead and make it an experimental
>>>>> RFC_.
>>>>> This would support the development of interoperable
>>>>> implementations, the
>>>>> collection of deployment data and also hopefully open source code.
>>>>> Though I have full faith in ICE, following the practices that
>>>>> made the
>>>>> IETF successful take precedent here, I believe.
>>>>> This is a change from the attached that I wrote on 7/16/2007.
>>>>> Note: We can only hope the proliferation of SBC in VoIP service
>>>>> provider
>>>>> networks will not make these concerns and ICE for SIP a moot
>>>>> issue, but
>>>>> this is another topic.
>>>>
>>>> Henry,
>>>>
>>>> I don't quite understand why you are requesting a status change of
>>>> ICE? To me it appears that ICE is almost finished spec. It has
>>>> been implemented, used and feedback has been influencing the
>>>> protocol. It is definitly one of the better examples of current
>>>> IETF work that actually has running code. Sure, the
>>>> implementations may not be fully updated yet to the latest draft
>>>> version. But I think it is safe to say that ICE works in the
>>>> intended core cases. If there are some corner cases where it
>>>> fails, that may be. But that is not information we will have until
>>>> really large scale deployement or someone makes very extensive =20
>>>> tests.
>>>
>>> Henry has, I believe, been very consistent in urging the IETF to
>>> have more "running code" experience on every specification (not
>>> just ICE) than our process requires. He has, IIRC, also advocated
>>> changing the process to require running code before publishing an
>>> RFC. He's even suggested at times that one should have running code
>>> before submitting a protocol draft.
>>>
>>> Another point to consider -- I've known Henry to bark at a tree
>>> with no squirrel in it occasionally, but most of the time he's
>>> pretty reliable, more so than most technology predictors. If he's
>>> worried about ICE, it's probably because he's been talking to
>>> people who are trying to implement it and they're telling him it is
>>> difficult or that they're having performance or reliability issues.
>>> I imagine that there are plenty of internal development teams in
>>> various companies that are working on this and not necessarily
>>> bringing their results back to the IETF. After all, if everybody
>>> sees the emperor as fully clothed, they'd be embarrassed to
>>> proclaim otherwise. Of course, I wish all of these developers were
>>> tied into the IETF, but then I'm sure I don't want to see every ice-
>>> implementor's question on the SIP list. That is why we have a sip-
>>> implementors mailing list, right?
>>>
>>> Personally, I think ICE is the best solution we have. We KNOW of
>>> cases where hole-punching doesn't work, and I think ICE will
>>> produce usable results in most of those cases. But that's just my
>>> personal opinion from reading the specs. I haven't implemented ICE,
>>> and I haven't used anybody else's implementation. Sure, it's
>>> complex, but that's because the underlying problem is complex --
>>> there are just too many different behaviors out there in "NAT and
>>> firewall land". Of course, I could be wrong, too. It's happened
>>> before, once or twice ;-). Just how sure are we that we're right
>>> about this?
>>>
>>> Are we designing a "spruce goose" or a practical system? Only
>>> experience will tell us. But some good flight reports from the
>>> prototypes could go a long way towards reassuring us of the
>>> design's utility.
>>>
>>> So far we've had a couple of people report hearsay evidence -- that
>>> somebody else has an ICE implementation and it works. Any of you
>>> folks have personal experience you can share? Have there been
>>> prototypes exercised at the interop events?
>>>
>>> --
>>> Dean
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Behave mailing list
>>> Behave@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/behave
>>
>>
>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
>
>
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Thu Jul 19 10:54:30 2007
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At 14:29 19/07/2007, Jan Janak wrote:
>Juha Heinanen wrote:
>> a while back i read ice tutorial slides and my impression was that the
>> protocol was very complex (pity implementers) and provided only a bit
>> better nat traversal capability than what [open]ser/rtpproxy/mediaproxy
>> have been offering to lots of users for years already.
>
> The NAT traversal algorithm implemented in [open]ser, rtpproxy, and
> mediaproxy had been developed long before the first version of the ICE
> draft; at that time no other solution existed (except for preliminary
> STUN and TURN drafts), and we needed something to make it work; in
> addition to that NAT traversal support in UA was virtually non-existent
> back then.
>
> Having said that, it is true that the algorithm implemented in
> [open]ser is simpler than ICE, but I believe that these two cannot
> be directly compared. Even the NAT traversal logic in SER can get
> pretty complex if none of the end-points is a PSTN gateway in the
> public internet (especially if you want to handle re-INVITEs and
> serial forking properly).

I don't think it is a question whether ICE is complex or not. The question
is if the complexity is worth it. An argumentation line one can choose is
the e2e principle which justifies the load of software loaded in the
end-devices (think for example of TCP) to gain reliability.

If ICE can accomplish the reliability promise, it also helps to migrate away 
from media relays (which is what SER does using rtpproxy) , which is a good thing.

So I think that the right course of action is that folks doubting ICE reliability
make their case (by that I mean specific arguments rather than emotional debates 
about red-colored nails and other hardware).

-jiri




--
Jiri Kuthan            http://iptel.org/~jiri/



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Subject: [Sip] RFC3891 (Replaces header) discrepancies
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It seems that there is discrepancy between a statement in RFC3891 and
some vendor implementations.

=20

   RFC3891, section 4 states:

=20

   A UAC MUST NOT send an INVITE with a Replaces header field that
attempts

   to replace an early dialog which was not originated by the target of
the

   INVITE with a Replaces header field.

=20

The issue appears to be related to the restriction of "originated" in
the above statement. It tends to imply that it is invalid if the early
dialog was "terminated" by the target rather than "originated" by the
target.

=20

Consider an attended transfer in which call1 is established between a
transferee and a transferor. The transferor creates a new call2 towards
transfer target. While this call is still alerting (early dialog), it
sends a REFER on call1 with a Replaces header that references call2. In
this case, the Replaces header attempts to replace an early dialog which
was "terminated" by the target.

=20

Thanks !

James


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>It seems that there is discrepancy between a =
statement in
RFC3891 and some vendor implementations.<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 style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; =
RFC3891,
section 4 states:<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><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 style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; A UAC =
MUST NOT
send an INVITE with a Replaces header field that =
attempts<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; to =
replace an
early dialog which was not originated by the target of =
the<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"'>&nbsp;&nbsp; INVITE with a Replaces header =
field.<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=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The issue appears to be related to the restriction of =
&#8220;originated&#8221;
in the above statement. It tends to imply that it is invalid if the =
early
dialog was &#8220;terminated&#8221; by the target rather than =
&#8220;originated&#8221;
by the target.<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'>Consider an attended transfer in which call1 is =
established
between a transferee and a transferor. The transferor creates a new =
call2
towards transfer target. While this call is still alerting (early =
dialog), it
sends a REFER on call1 with a Replaces header that references call2. In =
this
case, the Replaces header attempts to replace an early dialog which was =
&#8220;terminated&#8221;
by the target.<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'>Thanks !<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'>James<o:p></o:p></span></font></p>

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Subject: RE: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 


   a. extensive technical review, comments and feedbacks, including 
commentary from implementors (including one in my own company, and 
several others from outside of it) and deployers (Tim Moore from Microsoft)

FM: While I applaud the time and effort that the large vendors have
obviously invested in this, this is a qualitative statement.  Perhaps Cisco
and/or Microsoft have empirical data they would be willing to share about
robustness and performance?  I don't think anyone here, including myself, is
saying that ICE should not go forward or be used (its ability to handle
hairpinning is of great interest for example), we are asking for evidence
that we're not going to have another 3489 experience if we can avoid it.

   e. multiple open source implementations (two have been pointed out 
already)

FM: I think I missed those, can they be repeated?  The only one I'm aware of
is pjnath and its first release was a little over two months ago and its
based on -15.


Thanks,
FM




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Subject: RE: [P2PSIP] Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before
	LCfor RFC
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
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Phil,

You are right again, but as mentioned, concept papers (a true torrent
over many years lately) and discussion lists cannot replace published
measurement tools, measured results and tweaking the approach based on
measurements; measuring the improvements, etc.=20

This is the reason why I am not going into debating points about the
merits of ICE, but rather the best approach to make it successful and be
able to prove it.

ICE is not a protocol agreement like SIP, but a technology that faces
the great physical complexity on the net, so the protocol discussion
approach is no replacement for testing and tweaking. This probably
explains why we are in this discussion about ICE in the first place.

By the same token I-D process arguments and consensus are no replacement
either for proof on the net.=20

Henry

-----Original Message-----
From: Philip Matthews [mailto:philip_matthews@magma.ca]=20
Sent: Wednesday, July 18, 2007 8:19 PM
To: Henry Sinnreich
Cc: Magnus Westerlund; behave@ietf.org; mmusic@ietf.org; sip@ietf.org;
Pyda Srisuresh; Frank W. Miller; p2psip@ietf.org
Subject: [P2PSIP] Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before
LCfor RFC


On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:

> Phil,
>
>> In my view, ICE _is_ a "Hole Punching Technique".
>
> Yes, you absolutely right, though the devil is in the details.
>
> We cannot go here into a shoot-out between hole punching or other
> mentioned on the list vs. ICE, but there seems to be a huge difference
> in effectiveness and performance.

What precise algorithm are you talking about when you say this?
As far as I am concerned, ICE is simply a well-documented hole =20
punching algorithm.
It differs from others that I have seen written up in the following =20
ways:
- ICE has the concept of peer reflexive addresses, which other hole-=20
punching techniques that I have seen do not;
- ICE has the concept of controlling/controlled endpoint, which gives =20
the controller additional flexibility in choosing a path;
- ICE does NOT have the concept of port prediction, which some other =20
techniques do (e.g., Saikat's TCP work).

Philip

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On 7/19/07 10:19 AM, Frank W. Miller wrote:
> FM: While I applaud the time and effort that the large vendors have
> obviously invested in this, this is a qualitative statement.  Perhaps Cisco
> and/or Microsoft have empirical data they would be willing to share about
> robustness and performance?

That's the kind of data you get between "proposed standard" and "draft
standard" -- not *prior* to publication. I'm not sure why there's so
much noise about setting the bar much, much higher for ICE than we do
for other IETF proposals.

To be clear: to get this kind of experience (especially as it pertains
to interoperability), you need to have a stable (i.e., frozen)
specification. Internet Drafts simply don't provide an appropriate basis
for the kinds of proof that you and others are demanding.

I think we would all benefit from a quick review of RFC 2026:

>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.
...
>    Implementors should treat Proposed Standards as immature
>    specifications.  It is desirable to implement them in order to gain
>    experience and to validate, test, and clarify the specification.
>    However, since the content of Proposed Standards may be changed if
>    problems are found or better solutions are identified, deploying
>    implementations of such standards into a disruption-sensitive
>    environment is not recommended.


/a


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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides
	for body (MIME) handling)
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Hi,

yes, I agree there is a bigger issue with 'render'. Even though this 
bigger issue does not directly relate to multipart, we can also tackle 
it in the same draft and clarify how 'render' is used and what it means.

Cheers,

Gonzalo


Paul Kyzivat wrote:
> 
> 
> Gonzalo Camarillo wrote:
>> Hi Paul,
>>
>>> Regarding Content-Disposition in multiparts:
>>>
>>> While it seems "wrong" to me, I can live with using "render" as the 
>>> normal disposition for multipart, *if* we can do a careful job of 
>>> defining what "render" actually means for SIP. Used this way it 
>>> clearly doesn't always mean to display the content to a user.
>>
>> An entity receiving a 'multipart/mixed' body will process its body 
>> parts based on their disposition types.If the disposition type of a 
>> particular body part is not understood, it will be displayed to the 
>> user because the default disposition type is 'render'.
> 
> There is a bigger problem with 'render' than just multipart.
> 
> We have many uses of body parts that never mention Content-Disposition 
> at all. The related examples don't use it, and so they all get a default 
> of 'render'. This is particularly the case with events. E.g.
> 
> - the notification for REFER contains type application/sipfrag.
>   Its probably not intended to be "rendered" in any literal sense,
>   though some conclusion about success or failure may be.
> 
> - usage of filter bodies in presence subscriptions (RFC 4660)
>   doesn't have a specified disposition, so it implicitly uses render.
>   Its hard for me to convince myself that what the presence server
>   does with these can be described as rendering.
> 
> - In fact, the only place I can think of where 'render' is (implicitly)
>   used in sip and has the appropriate connotation is in MESSAGE.
> 
> I think we just have to grandfather the use of 'refer' in SIP, and 
> document that the usage in sip does not mean "render", but rather means 
> that the usage must be determined by other context - specifically the 
> type of message or response and the content type. IOW, 'render' in sip 
> can be thought of as a peculiar spelling for 'by-context'. It might be 
> helpful to give guidance for development of future specs about when 
> 'render' should be used vs defining a new disposition.
> 
>> Therefore, the 'multipart/mixed' body will never be processed based on 
>> its disposition type.
> 
> Based on my discussion above, I would consider this a case of the 
> multipart being processed according to its context *because* its 
> content-disposition is 'refer'. If it had some other content-disposition 
> then that might mean that it should be processed specially. E.g. if a 
> multipart/mixed had disposition 'by-reference' then that needs to be 
> taken into account, and the processing of the multipart becomes 
> conditional on the reference.
> 
> Also, there are only certain contexts in which multipart should be 
> processed "normally". Consider a NOTIFY from a REFER, with a 
> Content-Type of message/sipfrag, which happened to include a body as 
> part of the fragment. (This is legal.) That body might contain a 
> multipart/mixed, and that might contain other stuff. But I think the 
> processing of all of that is dependent on processing of the sipfrag. For 
> instance, if something inside that had handling=required, I don't think 
> that means that processing of the NOTIFY fails because the recipient 
> can't handle that.
> 
>> This means that it does not make much sense to talk about its 
>> disposition type as a whole.
> 
> Counterexample above.
> 
>> However, we need to have one in order to be able to set its handling 
>> parameter. That's why we can either keep 'render' explaining that it 
>> is actually never used or define a new disposition type.
> 
> I can go either way on that. But because there is existing usage, and 
> because 'render' semantics are already so messed up, I think it makes 
> more sense to stick with 'render' and just clarify its meaning.
> 
>> Since existing implementations will not understand our newly defined 
>> disposition type, they will treat it as 'render' anyway... but the 
>> resulting behavior would be the same. That's why it may make sense to 
>> keep using 'render'.
> 
> Yup.
> 
>> On the other hand, semantically, it would make more sense to define a 
>> new disposition type.
>>
>> Also, if we decide to define the 'by-reference' disposition type (per 
>> a different email thread), it could be used here as well.
> 
> It could be used when there is actually a reference to it. But it would 
> not be an appropriate disposition to use when there is no reference to 
> the multipart.
> 
>     Thanks,
>     Paul
> 
>> Cheers,
>>
>> Gonzalo
>>
>>
>>
>>> Gonzalo Camarillo wrote:
>>>> Folks,
>>>>
>>>> here you have the slides I have put together for the body (MIME) 
>>>> handling presentation on Monday:
>>>>
>>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>>
>>>>
>>>> These slides relate to this draft:
>>>>
>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>>
>>>>
>>>> The slides discuss all the open issues that have been brought up in 
>>>> the list in the last months.
>>>>
>>>> Comments are welcome.
>>>>
>>>> Thanks,
>>>>
>>>> Gonzalo
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>
>>



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Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies
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   From: "Jackson, James" <james_jackson@labs.att.com>

   Consider an attended transfer in which call1 is established between a
   transferee and a transferor. The transferor creates a new call2 towards
   transfer target. While this call is still alerting (early dialog), it
   sends a REFER on call1 with a Replaces header that references call2. In
   this case, the Replaces header attempts to replace an early dialog which
   was "terminated" by the target.

That is not allowed by RFC 3891.  But suppose it was allowed.  Further
suppose that the transfer target rings for another 90 seconds (sending
an additional 180) and then the proxy that manages the transfer target
cancels the call, causing the transfer target to send a 487.  Who
receives the 180 and 487?

Dale


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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [P2PSIP] Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before
	LCfor RFC
Date: Thu, 19 Jul 2007 13:51:47 -0400
To: Henry Sinnreich <hsinnrei@adobe.com>
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I agree that it would be very nice to see more published data on
when ICE succeeds (and specifically when it succeeds in finding a  
direct path).
It would also be nice to see more open source implementations.

It used to be that the Routing area would hold a draft in the "almost  
done" state
until they had some implementation experience, and then publish as an  
RFC only
once they had that data. Is this what you are looking for?

- Philip


On 19-Jul-07, at 11:30 , Henry Sinnreich wrote:

> Phil,
>
> You are right again, but as mentioned, concept papers (a true torrent
> over many years lately) and discussion lists cannot replace published
> measurement tools, measured results and tweaking the approach based on
> measurements; measuring the improvements, etc.
>
> This is the reason why I am not going into debating points about the
> merits of ICE, but rather the best approach to make it successful  
> and be
> able to prove it.
>
> ICE is not a protocol agreement like SIP, but a technology that faces
> the great physical complexity on the net, so the protocol discussion
> approach is no replacement for testing and tweaking. This probably
> explains why we are in this discussion about ICE in the first place.
>
> By the same token I-D process arguments and consensus are no  
> replacement
> either for proof on the net.
>
> Henry
>
> -----Original Message-----
> From: Philip Matthews [mailto:philip_matthews@magma.ca]
> Sent: Wednesday, July 18, 2007 8:19 PM
> To: Henry Sinnreich
> Cc: Magnus Westerlund; behave@ietf.org; mmusic@ietf.org; sip@ietf.org;
> Pyda Srisuresh; Frank W. Miller; p2psip@ietf.org
> Subject: [P2PSIP] Re: [Sip] RE: [BEHAVE] Re: ICE deployment data  
> before
> LCfor RFC
>
>
> On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:
>
>> Phil,
>>
>>> In my view, ICE _is_ a "Hole Punching Technique".
>>
>> Yes, you absolutely right, though the devil is in the details.
>>
>> We cannot go here into a shoot-out between hole punching or other
>> mentioned on the list vs. ICE, but there seems to be a huge  
>> difference
>> in effectiveness and performance.
>
> What precise algorithm are you talking about when you say this?
> As far as I am concerned, ICE is simply a well-documented hole
> punching algorithm.
> It differs from others that I have seen written up in the following
> ways:
> - ICE has the concept of peer reflexive addresses, which other hole-
> punching techniques that I have seen do not;
> - ICE has the concept of controlling/controlled endpoint, which gives
> the controller additional flexibility in choosing a path;
> - ICE does NOT have the concept of port prediction, which some other
> techniques do (e.g., Saikat's TCP work).
>
> Philip
>
> _______________________________________________
> P2PSIP mailing list
> P2PSIP@ietf.org
> https://www1.ietf.org/mailman/listinfo/p2psip
>



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Hi Paul,

>  > The interaction with handling=required would be that if you are not able
>  > to find a reference to a 'by-reference' body part with
>  > handling=required, you report an error because you have not been able to
>  > process the body part.
> 
> Hmm. I guess that makes sense. It means that any time there is an
> optional header with a reference to a body part then the handling of
> that body part needs to be optional. But again I guess that makes good
> sense.

Not really. I create an extension that consists of a header field (which 
is optional as any other new header field) that points to a body of 
disposition type 'by-reference'. If I want to make sure that the UAS 
understands the extension, I set the body's handling parameter to required.

If the UAS does not understand the header field, it will not find any 
reference to the body. Therefore, it will not be able to process it and 
will return an error response.

Of course, this could also be done with an option-tag in a Require 
header field.

> Some interesting questions arise when body parts are being processed by
> middleboxes. (Maybe we aren't supposed to talk about that here.) The
> whole issue of handling=required gets messy there. The middlebox isn't
> the intended recipient, so the handling parameter doesn't really apply
> to it.

Well, this is true (or we want to think it is true) in SIP. But this is 
what RFC 3459 has to say about optional body parts:

    An OPTIONAL body part is one the sender doesn't care whether the
    receiving system delivers it or not.  A content gateway can silently
    delete such body parts if the receiving system cannot deliver the
    part.

Maybe we should add some clarifying text in the draft before B2BUA 
implementers decide to remove all optional body parts they get their 
hands on :o)

>  So presumably it processes whatever headers and body parts it
> likes. That is another motivation for making the decision about
> processing be explicit - so it can be selective.

yes, being explicit is always good.

Cheers,

Gonzalo



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> -----Original Message-----
> From: Christer Holmberg (JO/LMF) 
> [mailto:christer.holmberg@ericsson.com] 
> Sent: Thursday, July 19, 2007 3:11 AM
> To: Dan Wing; sip@ietf.org
> Cc: ietf-rtpsec@imc.org
> Subject: RE: [Sip] SIP Identity using Media Path
> 
> 
> Hi, 
> 
> >>"in order for the mechanism to work, SBC-type-of-entities 
> >>must permit 
> >>DTLS, TLS, ICE, or HIP messages to be exchanged in the media path."
> >> 
> >>A small question for clarification: at what point must this 
> >>exchange 
> >>(two-way, I assume) work? As soon as the UAS has received 
> >>the INVITE?
> > 
> >Yes.
> 
> A little more on this.
> 
> As we have discussed before, you may not have two-way connectivity
> before 200 OK.
>
> I am pretty sure what will not work with most SBCs.
>
> Now assume that you will have two-way connectivity before you 
> have even sent the SDP answer. 


I think you meant to say "now assume that you will _not_ ..."?


In the case of no media path, it would only be possible for the
authentication server, and the UA receiving the SIP request, to 
cryptographically validate the SIP headers and a very minor 
portion of the SDP (From:, Date:, and the header that we 
generate that copies the a=fingerprint attributes from the 
SDP, and the few other headers described in 
-wing-sip-identity-media).

This provides some protection against attackers, and works through
SBCs.  Specifically, this protects against attackers who were not
able to obtain a validly-signed SIP request.


However, only validating those SIP headers is inadequate if
an attacker has captured or otherwise acquired a validly-signed 
SIP request, modified the SDP (such that the signature is 
still valid), and re-injected that SIP request into a SIP 
proxy that forwards it to you.  In order to detect such a 
replay attack, you need to perform one of the media-path based 
tests that I describe in wing-sip-identity-media (DTLS-SRTP, 
HIP, ICE, TLS).


> Also, even if you don't have any SBCs, but you still have NATs, the
> calling user often needs to send something in order to open the NAT
> binding, and allow media plane traffic (including your identity
> exchange) from the remote end. If you use ICE, or stand-alone 
> STUN, you
> will have an open NAT binding, but if your terminal opens the network
> e.g. by sending dummy RTP packets, it will not be able to do 
> so until it
> has received the SDP answer.

* But if you haven't received the SDP answer, where are you sending
those dummy RTP packets (what is their destination IP address and 
UDP port)?

* If the NAT perform address-based filtering of incoming packets, 
it drop incoming UDP packets until the device behind the NAT
(the SIP UA, in our case) sends a packet to the IP address sending
the incoming packet.  The SIP UA would learn that IP address upon
receipt of the SDP answer.  Further, if the NAT is doing port
filtering of incoming packets, the SIP UA needs to know the UDP
port the external packet is arriving from, too (which also shows
up in the answer).  I point out the problem with port filtering
NATs because sending dummy UDP packets to the SBC's IP address 
is not sufficient for those kinds of NATs.  Similarly, if the 
NAT is symmetric (assigns a new UDP port for every new 
destination port or destination address the SIP UA sends to), 
sending dummy UDP packets won't be effective for symmetric
NATs, either.

> So, I also think your solution requires support of ICE/STUN, 
> and in the case of ICE it also requires that the SBCs 
> don't mess up the SDP so that ICE will not be used.

No, it does not.

For protection against simple From: spoofing, it is adequate to
simply validate the SIP headers and the signature.  For protection
against replay attacks, it is necessary for both SIP UAs (*) to
implement one of HIP, ICE, DTLS-SRTP, or TLS.  

  (*) The replay attack prevention and implementation of HIP, ICE
      (with the extension described in -wing-sip-identity-media),
      DTLS-SRTP, or TLS can be done by the SIP UA itself _or_ by 
      the SBC associated with the UA's signing domain -- such as 
      the Enterprise's SBC if the enterprise is signing the identity 
      (dwing@cisco.com, dwing@boeing.com); or, if the SIP UA's 
      identity is signed by a Service Provider, the SBC operated by
      the service provider).

-d


> Regards,
> 
> Christer


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Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
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> -----Original Message-----
> From: Philip Matthews [mailto:philip_matthews@magma.ca]
> Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> David Barrett's algorithm, if I understand it correctly, would work in
> this situation. (As far as I can tell, David's is a simplified version
> of ICE).

That's correct.  The major differences (so far as I know) between what I'm
doing and what ICE does include:

- Peer's needn't exchange SDP before attempting connection establishment
(rather, the rendezvous server just tells both sides the others' LAN/NAT
addresses and they try to connect immediately). 

- Paths are evaluated in parallel rather than sequentially (ie, I attempt to
contact you direct via the LAN and indirectly via your NAT at the same time,
and then pick the faster of the two)

- There's no support for TURN servers as my application doesn't require them
(though it would be extremely straightforward to add).

- Clients attempt NAT classification as a *purely advisory* function to
support debugging and feed into certain non-critical pairwise optimizations.
(Said another way, even if the classification is totally wrong, it'll still
work.)

- Clients use adaptive keepalives to a STUN server to keep their NAT/FW
"hole" open.  By "adaptive" I mean I use probing to determine highest safe
keepalive frequency of the NAT/FW.  (Incidentally, there's no requirement
that any two clients use the same STUN server, and indeed I intend to move
to enable a "distributed STUN service" using the peers themselves.)

Basically, I do the minimum necessary for my requirements, which I respect
are much easier than SIP's.  However, know that despite my system being much
easier, it has been *extremely difficult and time consuming* to get working,
despite dozens of iterations drawing upon a network of hundreds of thousands
of machines on which to test.  Thus I would suggest that if my solution took
as long as it did, and went through as many variations as it did, the
SIP/ICE implementers should be very wary of hardening anything too early
before it hits the real world.  Because the real world is really, really
strange and unpredictable, I can guarantee you that.

Phillip's asked me to put together a bit more precise overview of the system
I use and I'll try to get to that soon.

-david




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From sip-bounces@ietf.org Thu Jul 19 14:02:04 2007
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Just UDP so far, though I harbor ambitions of adding TCP to the mix so it'll
try a bit of everything.

> -----Original Message-----
> From: Pyda Srisuresh [mailto:srisuresh@yahoo.com]
> Sent: Thursday, July 19, 2007 10:39 AM
> To: David Barrett; 'Melinda Shore'; 'Philip Matthews'
> Cc: sip@ietf.org; 'Henry Sinnreich'; behave@ietf.org; mmusic@ietf.org
> Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> David,
> 
> 90% success in connection rate is very impressive. Are you refering to
> UDP, TCP
> or both? As you say, fallback such as relay is essential for 100%
> connectivity.
> One cannot use hole punching without a good fallback scheme.
> 
> regards,
> suresh
> 
> --- David Barrett <dbarrett@quinthar.com> wrote:
> 
> > > -----Original Message-----
> > > From: Melinda Shore [mailto:mshore@cisco.com]
> > >
> > > ICE makes me want to pound red-hot nails into my eye sockets.  That
> said,
> > > 90% really isn't nearly good enough for residential deployments.
> >
> > Incidentally, 90% is where my stats come out today -- this doesn't
> include
> > restricted-cone/symmetric and symmetric/symmetric penetration, which has
> > been prototyped but not added to the production system yet.  The hope is
> > this will get us much close to the 99% range, but it remains to be seen
> by
> > how much.
> >
> > Furthermore, I'd wager that for most applications (such as VoIP), even
> > routing 1 in 10 calls through a central relay isn't going to break the
> bank.
> > (Or for mega-discounts, use the full-cone and non-NAT'd clients (>30%)
> to
> > act as "free" relays for their peers.)
> >
> > Thus while NAT penetration using what I guess we're calling the "hole
> > punching technique" has been demonstrated to work at 90%+ on a very
> large
> > scale (hundreds of thousands of clients tested) we should expect that
> it's
> > possible to get even better still.  If ICE can show how it'd do better,
> > that'd be great.  But we can't tolerate ICE being any worse than that,
> and
> > thus before rushing ahead with standardizing an unproven ICE, we should
> just
> > take the time to prove it.
> >
> > -david
> >
> >
> 




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To: "'David R Oran'" <oran@cisco.com>
Subject: RE: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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Actually, I'm not trying to reopen the requirements debate.  I just want =
ICE
measured against the requirements that were agreed upon.

And I completely agree that perfect is the enemy of the good.  The only =
ally
of the good is reality, and I want to enlist her on our side.

-david

> -----Original Message-----
> From: David R Oran [mailto:oran@cisco.com]
> Sent: Thursday, July 19, 2007 7:23 AM
> To: David Barrett
> Cc: 'Cullen Jennings'; 'SIP'; 'Behave WG'; mmusic@ietf.org
> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for =
RFC
>=20
>=20
> On Jul 19, 2007, at 6:15 AM, David Barrett wrote:
>=20
> > Hi Cullen, I think you bring up lot of really great and detailed
> > questions.
> >
> > However, speaking for myself at least, answering those questions
> > doesn't
> > really address my concern.  All I want to know is:
> >
> > 	Does ICE work, and how well?
> >
> > What "works" means is entirely open to debate.
> It may have been in the past, but the debate I think is closed,
> because the requirements captured the definition of "works". If you
> want to re-open the requirements discussion, please be clear about =
that.
>=20
> > I've tossed out "high peer
> > connectivity" and "low connection setup time" as the most important
> > criteria, and I've put hard data behind those numbers as a stake in
> > the
> > ground for comparison, but I'm open to other metrics as well.
> >
> See the requirements. One not often mentioned but clearly a
> requirement that ICE works very hard to meet is "use the most direct
> path available".
>=20
> > But regardless of the metrics, I want somebody who uses it and
> > loves it and
> > bases their business upon its success to stand up and explain why =
it's
> > great.
> > And I want this person to provide real-world data, sampled on a wide
> > scale, proving that ICE "works" -- by any definition.
> >
> This is in fact an extremely important exercise, and precisely the
> activity which according to RFC2026 is what should be happening after
> approval as proposed standard, and before advancement to  draft
> standard.
>=20
> > Basically, I want ICE firmly grounded in wide-scale testing and
> > real-world
> > data, and answering the questions below doesn't bring us any closer
> > to that.
> > Whether it's level 10 or level 12 doesn't produce new data.  How its
> > algorithms differ from competing ones doesn't show that they're any
> > better
> > or worse.  Detailed proposals for change don't shed any light on
> > whether
> > those changes improve or worsen it, because its current state is
> > completely
> > unmeasured.
> >
> Showing how the algorithms work better, or work in more cases that
> are covered by the requirements does in fact show that it's better,
> with respect to the requirements. Again, you're arguing indirectly
> that we should re-open the requirements discussion, which I do not
> think would be terribly productive, but would support if there is in
> fact consensus to do so.
>=20
> > The most powerful thing the IETF can do to standardize the world
> > upon ICE is
> > prove that it works better than any alternative.  Because until =
that's
> > proven, we just need to take it as a matter of faith.
> >
> I do not think that is desirable. The perfect is the enemy of the
> good. ICE does not have to be better than anything else. It just has
> to be "good enough" and see wide adoption.
>=20
> That's my 2=A2 (as an individual, not as an IAB member, in case =
anybody
> cares).
>=20
> DaveO
> > -david
> >
> >> -----Original Message-----
> >> From: Cullen Jennings [mailto:fluffy@cisco.com]
> >> Sent: Wednesday, July 18, 2007 7:56 PM
> >> To: SIP; Behave WG; mmusic@ietf.org
> >> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC
> >> for RFC
> >>
> >>
> >> First I apologize for the cross post - this is a mmusic draft, I
> >> would appreciate replies on the mmusic list only.
> >>
> >> Few things I am looking for here  ....
> >>
> >> 1) specific reasons why this work is less appropriate for standards
> >> track than other work that IETF puts on the standards track. Keep =
in
> >> mind the first step of standards track is Proposed Standards which
> >> requires basically no interoperability, then it could move to Draft
> >> Standard where interoperability is shown and finally Full Standard =
if
> >> it turns out to be a good idea.  If you are going to post a reply =
to
> >> this point, make sure you have read RFC 2026.
> >>
> >> 2) I am interested in the specifics of problems that ICE fails to
> >> solve that the WG has not previous considered. There are lots of
> >> people that have implemented versions of it and think it is a good
> >> idea - if people are having some problems with it, I'd love to hear
> >> about details of things that are wrong. That it has a lot of =
messages
> >> in some use cases has been known for a long time. It used to have =
far
> >> more messages  - engineering was done to consider what was a
> >> reasonable amount for call set up times and a new design was done =
to
> >> bring the message to a level most people invovled with the design
> >> felt was reasonable. If there are specific other than these that =
are
> >> wrong and not considered in the design, now would be a good time to
> >> get them on the table.
> >>
> >> 3) If people think that Skype for example has a simpler solution =
that
> >> works better than ICE. Feel free to describe exactly what it is.
> >> Several people invovled with the ICE design have looked in extreme
> >> detail at Skype and others and so far none of them have come back
> >> with what we should do different than ICE.
> >>
> >> 4) ICE can be used in a mode where it only provided the reflexive
> >> candidate. People were pretty clear that this was far inferior to =
the
> >> modes where there are other candidates. If someone would like to
> >> provide exactly how hole punching works better than this very =
limited
> >> ICE technique, I'd be glad to hear the details. This has certainly
> >> been an topic that any could have brought up over the last several
> >> years. If someone has specifics, I suggest they get out a full
> >> proposal really soon. I am very doubtful that hole punching =
solution
> >> is fundamentally different than the type of hole punching ICE does.
> >>
> >> 5) Some people have suggested that if all NATs were behave =
complaint,
> >> then we would not need ICE. I strongly disagree. I think the only
> >> part of ICE we would not need is the TURN relays which are more or
> >> less an optional part. Again, specifics on what can be removed are
> >> welcome.
> >>
> >> ICE is far more complicated that anyone wants - the problem is, no
> >> one has come up with a proposal that looks like it comes anywhere
> >> close to meeting the requirements and if simpler. The design space
> >> has been extensively explored by many people. The raw data to
> >> understand what will work on what NATs have been painfully tested =
and
> >> documented in publications such as:
> >>
> >> http://nutss.gforge.cis.cornell.edu/pub/imc05-tcpnat.pdf
> >>
> >> http://tools.ietf.org/id/draft-jennings-behave-test-results-04.txt
> >>
> >> https://www.guha.cc/saikat/stunt-results.php?
> >>
> >> (Ford used to have a page too but it seems to be gone)
> >>
> >> I have a very open mind to considering better ways to do ICE - but =
I
> >> need specific suggestions or it is impossible for me to really =
think
> >> about if the suggestions represent a better path or not.
> >>
> >> I would love to see more open source ICE code (there is lots =
already)
> >> - I feel that this is an area where ICE is likely just table stakes
> >> and owning an proprietary ICE stack will not turn out to be a
> >> competitive advantage. Having open source implementations where the
> >> development cost was shared and everyone could get working products
> >> to market faster would be an advantage to companies. Clearly I =
expect
> >> the bulk of ICE implementations to not be open source but I still
> >> would be keen on doing what I can to contribute to an open source
> >> version as personally I feel that is in the interest of end users, =
my
> >> employer, and the IETF as a SDO.
> >>
> >> Cullen <with my individual hat on>
> >>
> >>
> >>
> >> On Jul 18, 2007, at 9:12 AM, Dean Willis wrote:
> >>
> >>>
> >>> On Jul 18, 2007, at 3:10 AM, Magnus Westerlund wrote:
> >>>
> >>>> Henry Sinnreich skrev:
> >>>>> Following some discussions and soul searching, the best
> >>>>> approach for
> >>>>> ICE-17 for LC would be to _go ahead and make it an experimental
> >>>>> RFC_.
> >>>>> This would support the development of interoperable
> >>>>> implementations, the
> >>>>> collection of deployment data and also hopefully open source =
code.
> >>>>> Though I have full faith in ICE, following the practices that
> >>>>> made the
> >>>>> IETF successful take precedent here, I believe.
> >>>>> This is a change from the attached that I wrote on 7/16/2007.
> >>>>> Note: We can only hope the proliferation of SBC in VoIP service
> >>>>> provider
> >>>>> networks will not make these concerns and ICE for SIP a moot
> >>>>> issue, but
> >>>>> this is another topic.
> >>>>
> >>>> Henry,
> >>>>
> >>>> I don't quite understand why you are requesting a status change =
of
> >>>> ICE? To me it appears that ICE is almost finished spec. It has
> >>>> been implemented, used and feedback has been influencing the
> >>>> protocol. It is definitly one of the better examples of current
> >>>> IETF work that actually has running code. Sure, the
> >>>> implementations may not be fully updated yet to the latest draft
> >>>> version. But I think it is safe to say that ICE works in the
> >>>> intended core cases. If there are some corner cases where it
> >>>> fails, that may be. But that is not information we will have =
until
> >>>> really large scale deployement or someone makes very extensive
> >>>> tests.
> >>>
> >>> Henry has, I believe, been very consistent in urging the IETF to
> >>> have more "running code" experience on every specification (not
> >>> just ICE) than our process requires. He has, IIRC, also advocated
> >>> changing the process to require running code before publishing an
> >>> RFC. He's even suggested at times that one should have running =
code
> >>> before submitting a protocol draft.
> >>>
> >>> Another point to consider -- I've known Henry to bark at a tree
> >>> with no squirrel in it occasionally, but most of the time he's
> >>> pretty reliable, more so than most technology predictors. If he's
> >>> worried about ICE, it's probably because he's been talking to
> >>> people who are trying to implement it and they're telling him it =
is
> >>> difficult or that they're having performance or reliability =
issues.
> >>> I imagine that there are plenty of internal development teams in
> >>> various companies that are working on this and not necessarily
> >>> bringing their results back to the IETF. After all, if everybody
> >>> sees the emperor as fully clothed, they'd be embarrassed to
> >>> proclaim otherwise. Of course, I wish all of these developers were
> >>> tied into the IETF, but then I'm sure I don't want to see every =
ice-
> >>> implementor's question on the SIP list. That is why we have a sip-
> >>> implementors mailing list, right?
> >>>
> >>> Personally, I think ICE is the best solution we have. We KNOW of
> >>> cases where hole-punching doesn't work, and I think ICE will
> >>> produce usable results in most of those cases. But that's just my
> >>> personal opinion from reading the specs. I haven't implemented =
ICE,
> >>> and I haven't used anybody else's implementation. Sure, it's
> >>> complex, but that's because the underlying problem is complex --
> >>> there are just too many different behaviors out there in "NAT and
> >>> firewall land". Of course, I could be wrong, too. It's happened
> >>> before, once or twice ;-). Just how sure are we that we're right
> >>> about this?
> >>>
> >>> Are we designing a "spruce goose" or a practical system? Only
> >>> experience will tell us. But some good flight reports from the
> >>> prototypes could go a long way towards reassuring us of the
> >>> design's utility.
> >>>
> >>> So far we've had a couple of people report hearsay evidence -- =
that
> >>> somebody else has an ICE implementation and it works. Any of you
> >>> folks have personal experience you can share? Have there been
> >>> prototypes exercised at the interop events?
> >>>
> >>> --
> >>> Dean
> >>>
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Behave mailing list
> >>> Behave@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/behave
> >>
> >>
> >> _______________________________________________
> >> Behave mailing list
> >> Behave@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/behave
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip



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To: "'Jiri Kuthan'" <jiri@iptel.org>, "'Jan Janak'" <jan@iptel.org>,
	"'Juha Heinanen'" <jh@tutpro.com>
Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment databefore LC
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> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment databefore
> LC for RFC
> 
> So I think that the right course of action is that folks doubting ICE
> reliability
> make their case (by that I mean specific arguments rather than emotional
> debates
> about red-colored nails and other hardware).

Er, my "specific argument" is "until it's been proven to work, it hasn't
been proven to work".  It's tautological, but no less true for it.

Indeed, if you're *not* doubting ICE, why aren't you?

-david 




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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> ICE was built for reliability - to cover lots of cases and achieve not
> just 90% effectiveness (not nearly good enough), but as effective as is
> possible within the limitations of hurting the Internet or violating
> security policies and other things we won't touch.

Just to clarify, the 90% number (which I'm guessing comes from my published
stats) is the success rate of any two peers being able to establish a direct
connection without a relay service.  If I were to add a relay service (which
I don't need for my application, but would be trivial) for the remaining 10%
I hope you agree that near 100% of peers would be able to talk.  (I say
"near" because nothing works at 100% in the real world, no matter how simple
it appears on paper.)  


>    a. extensive technical review, comments and feedbacks, including
> commentary from implementors (including one in my own company, and
> several others from outside of it) and deployers (Tim Moore from
> Microsoft)

Excellent!  Do you or any of your implementor colleagues have any data
showing how well the result works?  Or if not, do you have an active system
that you can instrument to get it?  Or a test implementation that can be
instrumented in the lab?  Or any data at all showing how well ICE works?

I understand we have data about the general NAT penetration techniques,
years of experience, and active security teams.  Does anybody have actual
users, using this on a daily basis on a broad scale, proving that the
product of all this hard work accomplishes its goals?

-david





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From sip-bounces@ietf.org Thu Jul 19 14:08:38 2007
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Subject: RE: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
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> -----Original Message-----
> From: Adam Roach [mailto:adam@estacado.net]
> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
> 
> To be clear: to get this kind of experience (especially as it pertains
> to interoperability), you need to have a stable (i.e., frozen)
> specification. Internet Drafts simply don't provide an appropriate basis
> for the kinds of proof that you and others are demanding.

It seems premature to freeze the specification before we've proven that what
specified actually works.  Indeed, the most successful real-world projects
I've participated in worked in the exact opposite manner.

-da



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Subject: RE: [Sip] SIP Identity using Media Path
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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=20
Hi,
=20
>>>>"in order for the mechanism to work, SBC-type-of-entities must=20
>>>>permit DTLS, TLS, ICE, or HIP messages to be exchanged in the media=20
>>>>path."
>>>>=20
>>>>A small question for clarification: at what point must this exchange

>>>>(two-way, I assume) work? As soon as the UAS has received the=20
>>>>INVITE?
>>>=20
>>>Yes.
>>=20
>>A little more on this.
>>=20
>>As we have discussed before, you may not have two-way connectivity=20
>>before 200 OK.
>>
>>I am pretty sure what will not work with most SBCs.
>>
>>Now assume that you will have two-way connectivity before you have=20
>>even sent the SDP answer.
>=20
>=20
>I think you meant to say "now assume that you will _not_ ..."?

What I meant to say was: Now you assume that you will have two-way
connectivity before you have even sent the SDP answer :)


>In the case of no media path, it would only be possible for the
>authentication server, and the UA receiving the SIP request, to=20
>cryptographically validate the SIP headers and a very minor=20
>portion of the SDP (From:, Date:, and the header that we=20
>generate that copies the a=3Dfingerprint attributes from the=20
>SDP, and the few other headers described in -wing-sip-identity-media).
>=20
>This provides some protection against attackers, and works through
>SBCs.  Specifically, this protects against attackers who were not
>able to obtain a validly-signed SIP request.
>=20
>=20
>However, only validating those SIP headers is inadequate if
>an attacker has captured or otherwise acquired a validly-signed=20
>SIP request, modified the SDP (such that the signature is=20
>still valid), and re-injected that SIP request into a SIP=20
>proxy that forwards it to you.  In order to detect such a=20
>replay attack, you need to perform one of the media-path based=20
>tests that I describe in wing-sip-identity-media (DTLS-SRTP,=20
>HIP, ICE, TLS).
>=20
>=20

>>Also, even if you don't have any SBCs, but you still have NATs, the
>>calling user often needs to send something in order to open the NAT
>>binding, and allow media plane traffic (including your identity
>>exchange) from the remote end. If you use ICE, or stand-alone=20
>>STUN, you will have an open NAT binding, but if your terminal opens=20
>>the network e.g. by sending dummy RTP packets, it will not be able to
do=20
>>so until it has received the SDP answer.
>=20
>* But if you haven't received the SDP answer, where are you sending
>those dummy RTP packets (what is their destination IP address and=20
>UDP port)?

That is what I am saying: the user needs to get the SDP answer, so that
it can open the NAT binding (and allow incoming traffic)

>* If the NAT perform address-based filtering of incoming packets,=20
>it drop incoming UDP packets until the device behind the NAT
>(the SIP UA, in our case) sends a packet to the IP address sending
>the incoming packet.  The SIP UA would learn that IP address upon
>receipt of the SDP answer.  Further, if the NAT is doing port
>iltering of incoming packets, the SIP UA needs to know the UDP
>port the external packet is arriving from, too (which also shows
>up in the answer).  I point out the problem with port filtering
>NATs because sending dummy UDP packets to the SBC's IP address=20
>is not sufficient for those kinds of NATs.  Similarly, if the=20
>NAT is symmetric (assigns a new UDP port for every new=20
>destination port or destination address the SIP UA sends to),=20
>sending dummy UDP packets won't be effective for symmetric
>NATs, either.
>=20
>>So, I also think your solution requires support of ICE/STUN,=20
>>and in the case of ICE it also requires that the SBCs=20
>>don't mess up the SDP so that ICE will not be used.
>=20
>No, it does not.
>=20
>For protection against simple From: spoofing, it is adequate to
>simply validate the SIP headers and the signature.  For protection
>against replay attacks, it is necessary for both SIP UAs (*) to
>implement one of HIP, ICE, DTLS-SRTP, or TLS. =20

Ok, so let me put it this way: any feature in your solution that, in a
NAT requirement, does require end-to-end connectivity before the SDP
answer has been sent requires at least STUN (not necessarily ICE) - or
some other mechanism which the NAT binding is opened without having the
remote SDP. And, if you use ICE, the SBCs must not modify the SDPs in a
way so that ICE is disabled.

Regards,

Christer



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Hi Jonathan,

> The draft makes support of multipart a SHOULD in RFC 3261. If our 
> objective is to use multipart as a vehicle for extending SIP (e.g., by 
> including optional bodies in addition to the mandatory one - a location 
> object plus SDP for example), this is much better served by a MUST. 

I sent an email with the slides I intend to use to present this draft 
right before you sent your comments to the list. In those slides I 
already propose to make it a MUST. So, we are on the same page.

> Otherwise, a UAC will have to send a request first with multipart, and 
> if its rejected, retry the request with just the supported parts. Its 
> effectively the same as the behavior we'd see if the only way to extend 
> SIP was with Require, and not Supported.
> 
> This draft also strikes me as something that is in the essential 
> corrections category, but its debatable.
> 
> For multipart/alternative - I don't understand the requirement to order 
> them from least to most 'rich'.

It is not our requirement. It is just how MIME works.

> Is that a general MIME behavior or are 
> we defining it that way? If we want to prioritize, shouldn't we use an 
> explicit q-value or something? Using order seems like it will be brittle 
> (as it has been in SDP where m-lines were never numbered).

I do not think we should be changing how MIME works.

> The use case described for multipart/alternative - new SDP mechanisms - 
> seems kind of bogus. We've decided not to pursue that. Do we have a REAL 
> use case for this?

AFAICT, we have decided not to pursue SDPng but we still need a 
mechanism in order to be able to, at some point, migrate to new session 
description formats. The current consensus, as I understand it, is that 
multipart/alternative should not be used to provide alternative SDP 
descriptions, but that it could be used to provide alternative session 
descriptions written in different formats.

>> The reason for not having such a mechanism is that disposition types
>>    are typically supported within a context.  Outside that context, a UA
>>    (User Agent) may not support the disposition type. 
> 
> I think the same can be send for content-types. A UA may support a 
> content-type of application/sdp, but only for one disposition 
> ('session') and not another ('render'). So, generally I would say that a 
> UA never really supports a content-type; it supports a set of tuples of 
> {content-type, content-disposition, method}.

yes, that is what the draft intends to say. I will clarify it in the text.


>> UAs process message bodies and body parts in the following way.  On
>>    receiving a SIP message, if a body part is not referenced in any way
>>    (e.g., there are no header fields or other body parts with a
>>    Content-ID URL pointing to it), the UA processes the body part as
>>    indicated by its disposition type and the context in which the body
>>    part was received.
>>
>>    OPEN ISSUE 4: this means that a UA would need to parse all body parts
>>    to find references between them before being able to fully process
>>    them.  If we are OK with this, we need to include text here
>>    explaining it.
> 
> This behavior seems like a bad idea, and I don't understand why it is 
> needed. If the CID URL is part of an optional extension, then you make 
> the body handling optional, and the UA will know to pick up the body 
> part if it supports the extension, otherwise not.

we have a separate email thread where we are discussing this. The 
current proposal is to avoid having to process the message two times.

Thanks for your comments,

Gonzalo



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Gonzalo Camarillo wrote:
> Hi,
> 
> yes, I agree there is a bigger issue with 'render'. Even though this 
> bigger issue does not directly relate to multipart, we can also tackle 
> it in the same draft and clarify how 'render' is used and what it means.

Sounds good. Maybe its just a matter of changing the title of the draft 
to match the scope. Something like: 
draft-sipping-content-type-disp-clarification-nn.txt

	Paul

> Cheers,
> 
> Gonzalo
> 
> 
> Paul Kyzivat wrote:
>>
>>
>> Gonzalo Camarillo wrote:
>>> Hi Paul,
>>>
>>>> Regarding Content-Disposition in multiparts:
>>>>
>>>> While it seems "wrong" to me, I can live with using "render" as the 
>>>> normal disposition for multipart, *if* we can do a careful job of 
>>>> defining what "render" actually means for SIP. Used this way it 
>>>> clearly doesn't always mean to display the content to a user.
>>>
>>> An entity receiving a 'multipart/mixed' body will process its body 
>>> parts based on their disposition types.If the disposition type of a 
>>> particular body part is not understood, it will be displayed to the 
>>> user because the default disposition type is 'render'.
>>
>> There is a bigger problem with 'render' than just multipart.
>>
>> We have many uses of body parts that never mention Content-Disposition 
>> at all. The related examples don't use it, and so they all get a 
>> default of 'render'. This is particularly the case with events. E.g.
>>
>> - the notification for REFER contains type application/sipfrag.
>>   Its probably not intended to be "rendered" in any literal sense,
>>   though some conclusion about success or failure may be.
>>
>> - usage of filter bodies in presence subscriptions (RFC 4660)
>>   doesn't have a specified disposition, so it implicitly uses render.
>>   Its hard for me to convince myself that what the presence server
>>   does with these can be described as rendering.
>>
>> - In fact, the only place I can think of where 'render' is (implicitly)
>>   used in sip and has the appropriate connotation is in MESSAGE.
>>
>> I think we just have to grandfather the use of 'refer' in SIP, and 
>> document that the usage in sip does not mean "render", but rather 
>> means that the usage must be determined by other context - 
>> specifically the type of message or response and the content type. 
>> IOW, 'render' in sip can be thought of as a peculiar spelling for 
>> 'by-context'. It might be helpful to give guidance for development of 
>> future specs about when 'render' should be used vs defining a new 
>> disposition.
>>
>>> Therefore, the 'multipart/mixed' body will never be processed based 
>>> on its disposition type.
>>
>> Based on my discussion above, I would consider this a case of the 
>> multipart being processed according to its context *because* its 
>> content-disposition is 'refer'. If it had some other 
>> content-disposition then that might mean that it should be processed 
>> specially. E.g. if a multipart/mixed had disposition 'by-reference' 
>> then that needs to be taken into account, and the processing of the 
>> multipart becomes conditional on the reference.
>>
>> Also, there are only certain contexts in which multipart should be 
>> processed "normally". Consider a NOTIFY from a REFER, with a 
>> Content-Type of message/sipfrag, which happened to include a body as 
>> part of the fragment. (This is legal.) That body might contain a 
>> multipart/mixed, and that might contain other stuff. But I think the 
>> processing of all of that is dependent on processing of the sipfrag. 
>> For instance, if something inside that had handling=required, I don't 
>> think that means that processing of the NOTIFY fails because the 
>> recipient can't handle that.
>>
>>> This means that it does not make much sense to talk about its 
>>> disposition type as a whole.
>>
>> Counterexample above.
>>
>>> However, we need to have one in order to be able to set its handling 
>>> parameter. That's why we can either keep 'render' explaining that it 
>>> is actually never used or define a new disposition type.
>>
>> I can go either way on that. But because there is existing usage, and 
>> because 'render' semantics are already so messed up, I think it makes 
>> more sense to stick with 'render' and just clarify its meaning.
>>
>>> Since existing implementations will not understand our newly defined 
>>> disposition type, they will treat it as 'render' anyway... but the 
>>> resulting behavior would be the same. That's why it may make sense to 
>>> keep using 'render'.
>>
>> Yup.
>>
>>> On the other hand, semantically, it would make more sense to define a 
>>> new disposition type.
>>>
>>> Also, if we decide to define the 'by-reference' disposition type (per 
>>> a different email thread), it could be used here as well.
>>
>> It could be used when there is actually a reference to it. But it 
>> would not be an appropriate disposition to use when there is no 
>> reference to the multipart.
>>
>>     Thanks,
>>     Paul
>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>>
>>>
>>>> Gonzalo Camarillo wrote:
>>>>> Folks,
>>>>>
>>>>> here you have the slides I have put together for the body (MIME) 
>>>>> handling presentation on Monday:
>>>>>
>>>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>>>
>>>>>
>>>>> These slides relate to this draft:
>>>>>
>>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>>>
>>>>>
>>>>> The slides discuss all the open issues that have been brought up in 
>>>>> the list in the last months.
>>>>>
>>>>> Comments are welcome.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Gonzalo
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>>
>>>
> 


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From sip-bounces@ietf.org Thu Jul 19 14:30:04 2007
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Subject: RE: [Sip] RFC3891 (Replaces header) discrepancies
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From: "Jackson, James" <james_jackson@labs.att.com>
To: <Dale.Worley@comcast.net>, <sip@ietf.org>
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The assumption is that the transfer target rejects the initial early
dialog on receiving the REFER. The 180 and 487 would be sent in the
NOTIFY associated with the REFER.

I'm not trying to pick on any vendors here but take a look at what is
called "attended transfer with early completion" on this link:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_
guide09186a0080110c01.html

Thanks,
James

-----Original Message-----
From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20
Sent: Thursday, July 19, 2007 12:47 PM
To: sip@ietf.org
Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies

   From: "Jackson, James" <james_jackson@labs.att.com>

   Consider an attended transfer in which call1 is established between a
   transferee and a transferor. The transferor creates a new call2
towards
   transfer target. While this call is still alerting (early dialog), it
   sends a REFER on call1 with a Replaces header that references call2.
In
   this case, the Replaces header attempts to replace an early dialog
which
   was "terminated" by the target.

That is not allowed by RFC 3891.  But suppose it was allowed.  Further
suppose that the transfer target rings for another 90 seconds (sending
an additional 180) and then the proxy that manages the transfer target
cancels the call, causing the transfer target to send a 487.  Who
receives the 180 and 487?

Dale


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Just one comment in addition to what Gonzalo said:

Gonzalo Camarillo wrote:

>> The use case described for multipart/alternative - new SDP mechanisms 
>> - seems kind of bogus. We've decided not to pursue that. Do we have a 
>> REAL use case for this?
> 
> AFAICT, we have decided not to pursue SDPng but we still need a 
> mechanism in order to be able to, at some point, migrate to new session 
> description formats. The current consensus, as I understand it, is that 
> multipart/alternative should not be used to provide alternative SDP 
> descriptions, but that it could be used to provide alternative session 
> descriptions written in different formats.

The other classic example, because it is just like email, is MESSAGE 
with a multipart/alternative containing plain text and html.

	Paul


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On 7/19/07 1:08 PM, David Barrett wrote:
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@estacado.net]
>> Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
>>
>> To be clear: to get this kind of experience (especially as it pertains
>> to interoperability), you need to have a stable (i.e., frozen)
>> specification. Internet Drafts simply don't provide an appropriate basis
>> for the kinds of proof that you and others are demanding.
>>     
>
> It seems premature to freeze the specification before we've proven that what
> specified actually works.  Indeed, the most successful real-world projects
> I've participated in worked in the exact opposite manner.
>   

In that case, you're arguing with the IETF standards process, and have 
decided to beat on ICE as a proxy whipping boy for reasons that have no 
technical basis in ICE itself. The current process surrounding RFC 
publication is detailed in RFC 2026. Section 4.1 details the process 
you're misunderstanding. It's short. You should read it.

If you're up to it, RFC 3774 and RFC 3844 also provide some useful 
background on the work that has been done in this area recently.

Concerns about IETF RFC publication process in general -- and that's 
what you're arguing about here, not ICE itself -- are best addressed to 
the IETF-general mailing list and/or the IESG itself (as the conclusion 
of the problem WG was that issues surrounding the standards process were 
to be handed back to the IESG).

/a


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Hi Jeroen,

I don't see why 'ob' in the Contact is necessary in a response, but  
it doesn't hurt anything either.

thanks


On Jul 18, 2007, at 3:52 PM, Jeroen van Bemmel wrote:
> Minor comment on section 4.3 (sending requests): it talks about the  
> UAC inserting an 'ob' parameter in its Contact URI, should it do  
> the same for Contact URIs in responses over an outbound flow (for  
> consistency reasons perhaps; allows the other end to detect  
> outbound usage)?
>
> Regards,
> Jeroen
>
> Internet-Drafts@ietf.org wrote:
>>> 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 : Managing Client Initiated Connections in the Session
>>> Initiation Protocol (SIP) Author(s) : C. Jennings, R. Mahy
>>> Filename : draft-ietf-sip-outbound-10.txt
>>> Pages : 42
>>> Date : 2007-7-12
>>>
>>> The Session Initiation Protocol (SIP) allows proxy servers to
>>>   initiate TCP connections and send asynchronous UDP datagrams to
>>>   User Agents in order to deliver requests.  However, many practical
>>>   considerations, such as the existence of firewalls and Network
>>>   Address Translators (NATs), prevent servers from connecting to  
>>> User
>>>   Agents in this way.  This specification defines behaviors for User
>>>   Agents, registrars and proxy servers that allow requests to be
>>>   delivered on existing connections established by the User Agent.
>>>   It also defines keep alive behaviors needed to keep NAT bindings
>>>   open and specifies the usage of multiple connections.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-10.txt
>>>
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>>
>>
>>
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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [Sip] Re: [BEHAVE] Re: ICE deployment data before LC for RFC
Date: Thu, 19 Jul 2007 14:37:17 -0400
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On 19-Jul-07, at 11:38 , Adam Roach wrote:

> On 7/19/07 10:19 AM, Frank W. Miller wrote:
>> FM: While I applaud the time and effort that the large vendors have
>> obviously invested in this, this is a qualitative statement.   
>> Perhaps Cisco
>> and/or Microsoft have empirical data they would be willing to  
>> share about
>> robustness and performance?
>
> That's the kind of data you get between "proposed standard" and  
> "draft standard" -- not *prior* to publication. I'm not sure why  
> there's so much noise about setting the bar much, much higher for  
> ICE than we do for other IETF proposals.

Just to play the other side of the fence ...
In theory, RFCs advance from "proposed standard" to "draft standard"  
to "full standard".
In practice, this rarely happens.
In practice, once an RFC is published, it is either accepted and  
becomes a de-facto standard, or is ignored. Rarely is it revised and  
advanced along the standards track. And it is often difficult to  
discover which status (de-facto standard or ignored) a particular RFC  
actually has.
Look at SIP, for example. It is still at "proposed standard" and will  
likely stay there for a long time.

What Henry, David, and Frank want is some evidence that actually  
works (and thus will become the de-facto standard) before publishing  
the document as an RFC.

- Philip


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From sip-bounces@ietf.org Thu Jul 19 16:09:28 2007
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(as WG chair)

To all presenters on Monday and Tuesday.

If you are presenting slides in the SIP sessions, we would like these
posted to the meeting presentations site well before the meeting session
begins.

That means you need to get them to the chairs at least 24 hours in
advance of session start (and not 25 minutes after).

We will be projecting from what is on the meeting presentations site (no
exceptions) and therefore if your slides are not there, you will be
presenting to a blank screen.

Remember that as well as people trying to address your issues in the
meeting, we will have people trying to follow the discussion remotely.
Access to the slides in advance helps everybody.

And remember, we are looking for issues for the meeting to discuss, not
tutorials on the document contents - everyone is expected to have read
those.

Regards

Keith


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From sip-bounces@ietf.org Thu Jul 19 17:07:00 2007
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On 7/17/07 5:21 AM, DRAGE, Keith (Keith) wrote:
> (As WG chair)
>
> We reached the end of the time when WGLC comments were requested on
> Monday.
>
> If there are any last comments sitting out there that you have been
> meaning to submit, can I suggest you get them in an quickly as possible.
>   

Sorry for the late comment here, but I find section 3.3.2 perplexing in 
the current incarnation of the draft. It made sense before (e.g., in 
-02, which was the current version in Prague) -- but now that we're 
changing behavior, it seems completely out of place.

By my reading, section 3.3.2 basically says, "If you implement this 
specification, you are explicitly forbidden from doing the following 
procedure, which is now explained in enough detail to implement."

I would propose removing section 3.3.2.

/a


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My apology for the duplicate postings.=20
Having initiated this ample discussion about how to proceed with ICE-17,
it seems the following have emerged:

ICE-17 is an excellent document (bravo authors and contributors) and can
hardly get better at present.

The technical points discussed cannot change this fact either way.

Ironically, the defense of ICE-17 on procedural grounds makes even less
sense, since the culprit is not ICE-17, but the fact that present IETF
procedures do not make a clear enough distinction that everyone can
understand between I-D categories, such as:

* Concept memos; see the torrent in the P2PSIP WG
* Protocol standard agreements, such as for SIP
* Technology solutions such as ICE-17 or hole punching for NAT traversal
* I-Ds to support a business model (excuse me; I will not go there...)

Sorry to repeat: Process and consensus in meetings have nothing to do
with how good a technology may be. Only measurements and proof will do.

How does the IETF support the proof for a technology? (Yes, I have seen
all those quotes). Please remember that proving a technology also costs
money.
This is the reason why I trust more technologies coming out of funded
R&D, especially from University research. Vendors of technology may be
less likely to share such information.

Question: Is a standards organization the best place to develop and
promote new technology such as for NAT traversal? (all those p2p
companies and Skype did not think so...) Also, be prepared, SBC
traversal may come next.

In conclusion, there is nothing wrong with the ICE-17 paper IMHO, but
rather the IETF processes are either not satisfactory or not well
understood by folks like me who avoid reading process documents longer
than one web page :-)=20

I have taken therefore the liberty to copy <ietf@ietf.org> as Adam Roach
has suggested.

Thanks,=20
Henry Sinnreich





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> >I think you meant to say "now assume that you will _not_ ..."?
> 
> What I meant to say was: Now you assume that you will have two-way
> connectivity before you have even sent the SDP answer :)

Thanks for clarifying.

This is how we meet our requirement to avoid clipping, and meet
the requirement of RFC3264 to have a bi-directional path for
sendrecv streams:

      "Once the offerer has sent the offer, it MUST be prepared to
      receive media for any recvonly streams described by that offer.
      It MUST be prepared to send and receive media for any sendrecv
      streams in the offer, and send media for any sendonly streams in
      the offer (of course, it cannot actually send until the peer
      provides an answer with the needed address and port information)."

This requirement is captured in section 3.1 and requirement 5 (R5) of 
draft-wing-media-security-requirements-04.txt.

> >In the case of no media path, it would only be possible for the
> >authentication server, and the UA receiving the SIP request, to 
> >cryptographically validate the SIP headers and a very minor 
> >portion of the SDP (From:, Date:, and the header that we 
> >generate that copies the a=fingerprint attributes from the 
> >SDP, and the few other headers described in 
> -wing-sip-identity-media).
> > 
> >This provides some protection against attackers, and works through
> >SBCs.  Specifically, this protects against attackers who were not
> >able to obtain a validly-signed SIP request.
> > 
> > 
> >However, only validating those SIP headers is inadequate if
> >an attacker has captured or otherwise acquired a validly-signed 
> >SIP request, modified the SDP (such that the signature is 
> >still valid), and re-injected that SIP request into a SIP 
> >proxy that forwards it to you.  In order to detect such a 
> >replay attack, you need to perform one of the media-path based 
> >tests that I describe in wing-sip-identity-media (DTLS-SRTP, 
> >HIP, ICE, TLS).
> > 
> > 
> 
> >>Also, even if you don't have any SBCs, but you still have NATs, the
> >>calling user often needs to send something in order to open the NAT
> >>binding, and allow media plane traffic (including your identity
> >>exchange) from the remote end. If you use ICE, or stand-alone 
> >>STUN, you will have an open NAT binding, but if your terminal opens 
> >>the network e.g. by sending dummy RTP packets, it will not 
> be able to
> do 
> >>so until it has received the SDP answer.
> > 
> >* But if you haven't received the SDP answer, where are you sending
> >those dummy RTP packets (what is their destination IP address and 
> >UDP port)?
> 
> That is what I am saying: the user needs to get the SDP 
> answer, so that it can open the NAT binding (and allow 
> incoming traffic)

Yes, in cases with certain kinds of NATs which filter, it is 
necessary to receive the SDP answer.  In cases with certain
other kinds of NATs that don't filter (or where there is no
NAT), it isn't necessary to receive the SDP answer.

However, for reasons unrelated to SIP-Identity-Media, it is
important to verify the certificate in the SDP answer matches
the certificate exchanged over DTLS (or TLS, if doing comedia).
Otherwise, an attacker can send a well-timed DTLS ClientHello 
or TLS ClientHello which will fool the endpoint into exchanging
application-layer data (SRTP or TLS-encrypted application data)
with the attacker.  Such an exchange with an attacker is
undesirable.   Thus, to reduce call setup time, it's useful
to perform the DTLS/TLS exchange prior to receiving the SDP
answer.  But it isn't necessary to do so.  Certainly you
don't want to do anything with the data you receive until
you have validated the DTLS/TLS certificate.

> >* If the NAT perform address-based filtering of incoming packets, 
> >it drop incoming UDP packets until the device behind the NAT
> >(the SIP UA, in our case) sends a packet to the IP address sending
> >the incoming packet.  The SIP UA would learn that IP address upon
> >receipt of the SDP answer.  Further, if the NAT is doing port
> >iltering of incoming packets, the SIP UA needs to know the UDP
> >port the external packet is arriving from, too (which also shows
> >up in the answer).  I point out the problem with port filtering
> >NATs because sending dummy UDP packets to the SBC's IP address 
> >is not sufficient for those kinds of NATs.  Similarly, if the 
> >NAT is symmetric (assigns a new UDP port for every new 
> >destination port or destination address the SIP UA sends to), 
> >sending dummy UDP packets won't be effective for symmetric
> >NATs, either.
> > 
> >>So, I also think your solution requires support of ICE/STUN, 
> >>and in the case of ICE it also requires that the SBCs 
> >>don't mess up the SDP so that ICE will not be used.
> > 
> >No, it does not.
> > 
> >For protection against simple From: spoofing, it is adequate to
> >simply validate the SIP headers and the signature.  For protection
> >against replay attacks, it is necessary for both SIP UAs (*) to
> >implement one of HIP, ICE, DTLS-SRTP, or TLS.  
> 
> Ok, so let me put it this way: any feature in your solution that, in a
> NAT requirement, does require end-to-end connectivity before the SDP
> answer has been sent requires at least STUN (not necessarily ICE) - or
> some other mechanism which the NAT binding is opened without 
> having the
> remote SDP. And, if you use ICE, the SBCs must not modify the 
> SDPs in a way so that ICE is disabled.

draft-wing-sip-identity-media does not require end-to-end media 
path connectivity before the SDP answer arrives at the offer.

draft-wing-sip-identity-media provides identity protection in 
two "phases", to handle simple attacks and to handle more 
sophisticated attacks.  The first phase is the SIP header
signing, which is validated entirely without any regard to the
media path.  The second phase uses the media path, and protects
against replay attacks (where an attacker obtained a signed
Invite, for example, and replays it in order to ring your phone
again and appear to be the original caller.)  To protect against
that replay attack, you need to make the caller prove they know
something that only they know.  In sip-identity-media, I am
making the caller prove that he knows the private key associated
with the certificate (TLS, DTLS) or public key (HIP, ICE) that
is going to be used for the TLS, DTLS, or HIP call setup.  If 
he proves he knows that private key, it is safe and reasonable
to display the caller id.  If he can't prove that, he is an 
imposter.  If he refuses to prove that (or the media path is
blocked), you run the risk that -- after the media path is 
opened -- you will do the DTLS/HIP/TLS/ICE exchange and at
that time you will discover the user is an imposter.  That
discovery will occur after the phone has been rung and 
answered.  To me, that is a poor time to discover an 
imposter.  It's better than what we have today, though, so
I guess it _is_ forward progress and I could live with an
implementation that discovered an imposter after the phone
is answered.  It is certainly better than what we have 
today.  If such imposters become commonplace, users will
complain to their network providers about the middle-of-the-
night call alerting, phone answering, and attendant "Whoops,
sorry, that was an imposter.  We will send him a big fat
bill.  Please resume sleeping" messages displayed on their
handsets.

-d




> Regards,
> 
> Christer


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From sip-bounces@ietf.org Thu Jul 19 18:33:18 2007
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Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies
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   From: "Jackson, James" <james_jackson@labs.att.com>

   The assumption is that the transfer target rejects the initial early
   dialog on receiving the REFER. The 180 and 487 would be sent in the
   NOTIFY associated with the REFER.

I don't see what you're describing here.  If the INVITE/Replaces were
to complete, you would have the previous early dialog ended (probably
with 487).  And the INVITE/Replaces would presumably get a 200
quickly.  But the phone is still ringing.  To what "other end" UA does
it send provisional and final responses in regard to its status?

   I'm not trying to pick on any vendors here but take a look at what is
   called "attended transfer with early completion" on this link:

   http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_
   guide09186a0080110c01.html

>From the looks of Figure 4 on that page, the INVITE/Replaces does
*not* complete immediately, but rather its provisional and final
responses are the continuation of the previous early dialog at the
transer target.

I wonder how well that would interoperate.  Previously,
INVITE/Replaces completes quickly, but there is no reason that it
*must*, and transferee phones should be able to cope with long
transaction times for INVITE/Replaces.

Dale


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From sip-bounces@ietf.org Thu Jul 19 18:48:08 2007
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Subject: RE: [Sip] WGLC on draft-ietf-sip-sips
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You are absolutely correct: this is insane. Like giving matches to kids
and telling them not
to use it.

I don't recall who it was that wanted the procedures to remain in the
text. I believe the
rationale was "how to deal with legacy SIPS implementations".

I really think the text would be much simpler if we removed section
3.3.2 altogether.=20


> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]=20
> Sent: Thursday, July 19, 2007 14:07
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org
> Subject: Re: [Sip] WGLC on draft-ietf-sip-sips
>=20
> On 7/17/07 5:21 AM, DRAGE, Keith (Keith) wrote:
> > (As WG chair)
> >
> > We reached the end of the time when WGLC comments were requested on=20
> > Monday.
> >
> > If there are any last comments sitting out there that you have been=20
> > meaning to submit, can I suggest you get them in an quickly=20
> as possible.
> >  =20
>=20
> Sorry for the late comment here, but I find section 3.3.2=20
> perplexing in the current incarnation of the draft. It made=20
> sense before (e.g., in -02, which was the current version in=20
> Prague) -- but now that we're changing behavior, it seems=20
> completely out of place.
>=20
> By my reading, section 3.3.2 basically says, "If you=20
> implement this specification, you are explicitly forbidden=20
> from doing the following procedure, which is now explained in=20
> enough detail to implement."
>=20
> I would propose removing section 3.3.2.
>=20
> /a
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Thu Jul 19 19:15:56 2007
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Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>
Subject: [Sip] Outbound: TCP and SigComp
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With the move to CRLF for keepalive in outbound, the language discussing 
STUN handling with SigComp in TCP is also gone. This makes sense, since 
TCP no longer uses STUN.

However, there was no equivalent language added for how to handle the 
CRLF keepalives in conjunction with SigComp.

As with the earlier STUN mechanism, you can't just send the CRLF stuff 
down the TCP connection -- it needs to be wrapped in SigComp. Of course, 
we could employ the same mechanism originally proposed for STUN (the 
well-known shim), which would be the 13-byte SigComp shim, plus 2 bytes 
(CRLF) or 4 bytes (double CRLF).

Alternately, we can shave this down a _little_ bit by using bytecodes 
specifically designed to output just the CRLFs we need; for a 
double-CRLF, it would look like:

    at (128)
    OUTPUT (crlf, 4)
    JUMP(end)

    :crlf byte(13,10,13,10)

    :end
    END-MESSAGE (0,0,0,0,0,0,0)

Which translates to the following 14 bytes: f8 00 b1 22 a0 86 04 16 06 
0d 0a 0d 0a 23


(Yes, I realize this is 14 bytes instead of the 17 bytes you get with 
the shim, so it might not be worth using custom code).

For a single CRLF, such code would end up looking like this:

    at (64)
    :crlf pad(2)

    at (128)
    LOAD (crlf, 3338)
    OUTPUT (crlf, 2)
    END-MESSAGE (0,0,0,0,0,0,0)

Which translates to the following 11 bytes: f8 00 81 0e 86 ad 0a 22 86 
02 23.

In practice, we're probably best off just normatively saying that the 
CRLF characters are sent inside SigComp, and that the mechanism for 
representing them is a matter to be handled by the SigComp compressor at 
the sending end. We can point out the shim, and we can include the 
foregoing purpose-built SigComp messages, but there may be architectural 
complications to using either of them, so the ultimate requirement is 
simply to send something that produces "CRLF" as its output when it is sent.

/a


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From sip-bounces@ietf.org Thu Jul 19 20:05:09 2007
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Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>
Subject: [Sip] Outbound: Feature Tag (Ab)Use
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At the risk of further bloodying the corpse of this poor horse, I still 
think the usage of feature tags in outbound is way, way off in terms of 
the semantics described in 3261.

Reviewing the way this is defined to work in SIP:

3261:8.1.1.9:
>    If the UAC supports extensions to SIP that can be applied by the
>    server to the response, the UAC SHOULD include a Supported header
>    field in the request listing the option tags (Section 19.2) for those
>    extensions.

3261:8.2.4:

>    A UAS that wishes to apply some extension when generating the
>    response MUST NOT do so unless support for that extension is
>    indicated in the Supported header field in the request.  If the
>    desired extension is not supported, the server SHOULD rely only on
>    baseline SIP and any other extensions supported by the client.
> ...
>    Any extensions applied to a non-421 response MUST be listed in a
>    Require header field included in the response.  Of course, the server
>    MUST NOT apply extensions not listed in the Supported header field in
>    the request.  As a result of this, the Require header field in a
>    response will only ever contain option tags defined in standards-
>    track RFCs.

There are five normative statements in the text I quote above. The 
mechanism described in the outbound draft actually manages to violate 
_all_ _five_ of them.

The registrar infers *both* support for outbound *and* a request to 
apply the outbound extension by the presence of the "reg-id" Contact 
header field parameter. In particular, the client is apparently NOT 
supposed to send a "Supported: outbound" header field. This appears to 
contravene the normative statement in 8.1.1.9 cited above.

The registrar indicates that outbound procedures are being applied by 
including a "Supported" header field in the REGISTER response. (In 3261, 
"Supported" in a _response_ indicates that an extension _can_ _be_ 
supported in a future request, NOT that it is being applied to the 
current one -- sections 11 and 13.3.1.4 both talk about Supported usage 
in responses.)

In other words, the registrar (acting as a UAS) is applying an extension 
when generating a response without that extension being indicated in the 
"Supported" header field in the request. Again, this directly 
contravenes two normative statements from section 8.2.4 (cited above).

The registrar is further applying an extension to a non-421 response 
that is not listed in a "Require" header field, in violation of the 
remaining normative statement in the portion of section 8.2.4 cited above.

Why are we doing this so very, very wrong? It's almost as if we've gone 
out of our way to violate both the spirit AND the law of feature tag 
handling in as many ways as possible. Most annoyingly, we get exactly 
the desired behavior if we just do things the way they're described in 
3261. To wit:

    * If a client supports outbound, it includes "Supported: outbound"
      in all REGISTER requests, regardless of whether the specific
      request makes use of that extension.

    * The client adds a "reg-id" header field parameter to any Contacts
      to which outbound processing is to apply.

    * The registrar includes "Require: outbound" in any REGISTER
      requests to which outbound behavior is being applied.

    * The registrar is free to include "Supported: outbound" in any
      responses it generates -- and it means exactly what 3261 intends
      it to mean: that outbound is *supported*, but is not being applied
      to the response in question.

/a



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From sip-bounces@ietf.org Thu Jul 19 20:07:57 2007
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Rohan Mahy" <rohan@ekabal.com>,
	"Cullen Jennings" <fluffy@cisco.com>
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Subject: [Sip] draft-ietf-sip-outbound-10.txt : flow token construction
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Hi,

A comment / clarification on outbound-10 section 5.3 Forwarding requests, 
regarding flow token construction and incoming/outgoing logic:
The text says to send a 430 (Flow Failed) "For connection-oriented 
transports, if the flow no longer exists" (SHOULD)

This can be read as saying that if the original TCP connection no longer 
exists, a 430 should be sent. However, if the UAC somehow lost its 
connection and recovered, there would be a new TCP connection with (likely) 
a different ephemeral port at the UAC.

For the proposed flow token algorithm, this would cause the described 
incoming/outgoing logic to fail (since the token no longer matches an 
incoming flow from the UAC). Moreover, for requests in the other direction 
("incoming") the proxy would not be able to locate the new flow, and send a 
430 where it could have done better.

Two thoughts:
1. To determine if a (non-REGISTER) request is incoming/outgoing, it might 
be easier to simply say: 1 Via header --> outgoing, else incoming. There are 
other options, the Edge proxy could check the request URI for a Contact URI 
it has seen in REGISTER before. It could also be based on Service-Route 
and/or Path.

Perhaps this draft need not explain all details, but it currently only 
describes logic for mid-dialog requests, and appears to fail in the above 
described case of TCP connection loss.

2. An alternative implementation could be to maintain a mapping of Contact 
address ( IP/host+port ) to flow, which is updated each time a REGISTER 200 
OK comes by. If the UAC is using the same Contact URIs as in its REGISTER 
(or a GRUU rewritten by the home proxy), there is no need for a flow token 
at all: the Edge proxy can use (IP/host+port part of) the request URI as an 
(hash) index into the flow table (for incoming requests).

There are many other designs possible. Main point is that the stateless flow 
token approach described seems suboptimal in case of TCP connection loss 
(home proxy may be able to recover if the UAC used 2 or more flows, but if 
all were lost and the backup edge proxy also implemented this algorithm, 
ongoing dialogs would be lost)

Regards,
Jeroen



_______________________________________________
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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 sip-bounces@ietf.org Thu Jul 19 20:18:15 2007
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Adam Roach" <adam@nostrum.com>,
	"Rohan Mahy" <rohan@ekabal.com>
References: <46928F5F.3080502@softarmor.com> <469FFC30.2090309@nostrum.com>
Subject: Re: [Sip] Outbound: Feature Tag (Ab)Use
Date: Fri, 20 Jul 2007 02:15:10 +0200
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Good points, some nits inline

Regards,
Jeroen

Adam Roach wrote:
> At the risk of further bloodying the corpse of this poor horse, I
> still think the usage of feature tags in outbound is way, way off in
> terms of the semantics described in 3261.
>
> Reviewing the way this is defined to work in SIP:
>
> 3261:8.1.1.9:
>>    If the UAC supports extensions to SIP that can be applied by the
>>    server to the response, the UAC SHOULD include a Supported header
>>    field in the request listing the option tags (Section 19.2) for
>>    those extensions.
>
> 3261:8.2.4:
>
>>    A UAS that wishes to apply some extension when generating the
>>    response MUST NOT do so unless support for that extension is
>>    indicated in the Supported header field in the request.  If the
>>    desired extension is not supported, the server SHOULD rely only on
>>    baseline SIP and any other extensions supported by the client.
>> ...
>>    Any extensions applied to a non-421 response MUST be listed in a
>>    Require header field included in the response.  Of course, the
>>    server MUST NOT apply extensions not listed in the Supported
>>    header field in the request.  As a result of this, the Require
>>    header field in a response will only ever contain option tags
>>    defined in standards- track RFCs.
>
> There are five normative statements in the text I quote above. The
> mechanism described in the outbound draft actually manages to violate
> _all_ _five_ of them.
>
> The registrar infers *both* support for outbound *and* a request to
> apply the outbound extension by the presence of the "reg-id" Contact
> header field parameter. In particular, the client is apparently NOT
> supposed to send a "Supported: outbound" header field. This appears to
> contravene the normative statement in 8.1.1.9 cited above.
>
> The registrar indicates that outbound procedures are being applied by
> including a "Supported" header field in the REGISTER response. (In
> 3261, "Supported" in a _response_ indicates that an extension _can_
> _be_ supported in a future request, NOT that it is being applied to
> the current one -- sections 11 and 13.3.1.4 both talk about Supported
> usage in responses.)
>
> In other words, the registrar (acting as a UAS) is applying an
> extension when generating a response without that extension being
> indicated in the "Supported" header field in the request. Again, this
> directly contravenes two normative statements from section 8.2.4 (cited 
> above).
>
> The registrar is further applying an extension to a non-421 response
> that is not listed in a "Require" header field, in violation of the
> remaining normative statement in the portion of section 8.2.4 cited
> above.
> Why are we doing this so very, very wrong? It's almost as if we've
> gone out of our way to violate both the spirit AND the law of feature
> tag handling in as many ways as possible. Most annoyingly, we get
> exactly the desired behavior if we just do things the way they're 
> described in
> 3261. To wit:
>
>    * If a client supports outbound, it includes "Supported: outbound"
>      in all REGISTER requests, regardless of whether the specific
>      request makes use of that extension.
>

[jvb] this would probably be a MUST, right?

>    * The client adds a "reg-id" header field parameter to any Contacts
>      to which outbound processing is to apply.
>
>    * The registrar includes "Require: outbound" in any REGISTER
>      requests to which outbound behavior is being applied.

[jvb] you mean responses

>
>    * The registrar is free to include "Supported: outbound" in any
>      responses it generates -- and it means exactly what 3261 intends
>      it to mean: that outbound is *supported*, but is not being
>      applied to the response in question.

[jvb] is not _necessarily_ being applied, it depends on presence of Require: 
outbound (only). Guess this is a MAY, right?

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



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From sip-bounces@ietf.org Thu Jul 19 20:36:01 2007
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In-reply-to: <469FFC30.2090309@nostrum.com> (adam@nostrum.com)
Subject: Re: [Sip] Outbound: Feature Tag (Ab)Use
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   From: Adam Roach <adam@nostrum.com>
   Date: Thu, 19 Jul 2007 19:05:04 -0500
   Received-SPF: pass (nostrum.com: 75.53.54.121 is authenticated by a trusted
	   mechanism)
   X-Scan-Signature: 00e94c813bef7832af255170dca19e36
   Cc: sip@ietf.org, Cullen Jennings <fluffy@cisco.com>

   At the risk of further bloodying the corpse of this poor horse, I still 
   think the usage of feature tags in outbound is way, way off in terms of 
   the semantics described in 3261.

   Reviewing the way this is defined to work in SIP:

   3261:8.1.1.9:
   >    If the UAC supports extensions to SIP that can be applied by the
   >    server to the response, the UAC SHOULD include a Supported header
   >    field in the request listing the option tags (Section 19.2) for those
   >    extensions.

   3261:8.2.4:

   >    A UAS that wishes to apply some extension when generating the
   >    response MUST NOT do so unless support for that extension is
   >    indicated in the Supported header field in the request.  If the
   >    desired extension is not supported, the server SHOULD rely only on
   >    baseline SIP and any other extensions supported by the client.
   > ...
   >    Any extensions applied to a non-421 response MUST be listed in a
   >    Require header field included in the response.  Of course, the server
   >    MUST NOT apply extensions not listed in the Supported header field in
   >    the request.  As a result of this, the Require header field in a
   >    response will only ever contain option tags defined in standards-
   >    track RFCs.

   There are five normative statements in the text I quote above. The 
   mechanism described in the outbound draft actually manages to violate 
   _all_ _five_ of them.

   The registrar infers *both* support for outbound *and* a request to 
   apply the outbound extension by the presence of the "reg-id" Contact 
   header field parameter. In particular, the client is apparently NOT 
   supposed to send a "Supported: outbound" header field. This appears to 
   contravene the normative statement in 8.1.1.9 cited above.

   The registrar indicates that outbound procedures are being applied by 
   including a "Supported" header field in the REGISTER response. (In 3261, 
   "Supported" in a _response_ indicates that an extension _can_ _be_ 
   supported in a future request, NOT that it is being applied to the 
   current one -- sections 11 and 13.3.1.4 both talk about Supported usage 
   in responses.)

   In other words, the registrar (acting as a UAS) is applying an extension 
   when generating a response without that extension being indicated in the 
   "Supported" header field in the request. Again, this directly 
   contravenes two normative statements from section 8.2.4 (cited above).

   The registrar is further applying an extension to a non-421 response 
   that is not listed in a "Require" header field, in violation of the 
   remaining normative statement in the portion of section 8.2.4 cited above.

   Why are we doing this so very, very wrong? It's almost as if we've gone 
   out of our way to violate both the spirit AND the law of feature tag 
   handling in as many ways as possible. Most annoyingly, we get exactly 
   the desired behavior if we just do things the way they're described in 
   3261. To wit:

       * If a client supports outbound, it includes "Supported: outbound"
	 in all REGISTER requests, regardless of whether the specific
	 request makes use of that extension.

       * The client adds a "reg-id" header field parameter to any Contacts
	 to which outbound processing is to apply.

       * The registrar includes "Require: outbound" in any REGISTER
	 requests to which outbound behavior is being applied.

       * The registrar is free to include "Supported: outbound" in any
	 responses it generates -- and it means exactly what 3261 intends
	 it to mean: that outbound is *supported*, but is not being applied
	 to the response in question.

   /a



   _______________________________________________
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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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   From: Adam Roach <adam@nostrum.com>

   At the risk of further bloodying the corpse of this poor horse, I still 
   think the usage of feature tags in outbound is way, way off in terms of 
   the semantics described in 3261.

You're completely right, of course.

   Why are we doing this so very, very wrong? It's almost as if we've gone 
   out of our way to violate both the spirit AND the law of feature tag 
   handling in as many ways as possible.

And outbound is not the only place this has happened.  It took a long
time to purge this sort of error from gruu as well.  IMHO, I think the
reason is that people intuitively want "Supported" to mean "this
message is invoking the feature in question" (which is ambiguous as to
whether it means that the message contains use of the feature, or that
the message requests the far end to apply the feature).  People seem
to have a hard time grasping that its purpose is to communicate a
property of the UA, namely that the UA supports the feature in
question.  But many people in the world of software don't grasp
talking about "properties that things possess", they can only talk
about "actions happening right now".

Dale


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From: "vishal jogi" <vishaljogi@gmail.com>
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 Hi Vikram,

I have checked draft-ietf-simple-event-list-07, but it doesn't specify
anything about this case,
is there any draft or RFC which will give exact idea about it??

Thanks,
Vishal...

 On 7/19/07, Vikram Chhibber <vikram.chhibber@gmail.com> wrote:
>
> I don't think that this is correct behavior of the RLS. If the RLS
> understands the uri pointing to a resource-list. It should accept this
> subscription. Even though the list is empty. The user/administrator
> may provision entries in this list in future by non-SIP mechanism
> (XCAP).
>
> On 7/19/07, vishal jogi < vishaljogi@gmail.com> wrote:
> > Hi,
> > I have one doubt.
> >
> > i m subscribing to RLS uri,
> > which is refering one Resource-list which doesn't have any <entry>
> > mean it's blank <LIST>
> >
> > server is sending 404 as response for Subscribe, can anyone tell me it's
> > correct behaviour of server,
> > as that <LIST> is existing
> >
> > Thanks,
> > Vishal...
> > _______________________________________________
> > Sip mailing list
> > https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>

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<span class="gmail_quote"></span>
<div>Hi Vikram,</div>
<div>&nbsp;</div>
<div>I have checked draft-ietf-simple-event-list-07, but it doesn&#39;t specify anything about this case,</div>
<div>is there any draft or RFC which will give exact idea about it??</div>
<div>&nbsp;</div>
<div>Thanks,</div><span class="sg">
<div>Vishal...<br>&nbsp;</div></span>
<div><span class="e" id="q_113e11c976c7fcee_2">
<div><span class="gmail_quote">On 7/19/07, <b class="gmail_sendername">Vikram Chhibber</b> &lt;<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:vikram.chhibber@gmail.com" target="_blank">vikram.chhibber@gmail.com
</a>&gt; wrote:</span> 
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I don&#39;t think that this is correct behavior of the RLS. If the RLS<br>understands the uri pointing to a resource-list. It should accept this 
<br>subscription. Even though the list is empty. The user/administrator<br>may provision entries in this list in future by non-SIP mechanism<br>(XCAP).<br><br>On 7/19/07, vishal jogi &lt;<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:vishaljogi@gmail.com" target="_blank">
 vishaljogi@gmail.com</a>&gt; wrote:<br>&gt; Hi,<br>&gt; I have one doubt.<br>&gt;<br>&gt; i m subscribing to RLS uri,<br>&gt; which is refering one Resource-list which doesn&#39;t have any &lt;entry&gt;<br>&gt; mean it&#39;s blank &lt;LIST&gt; 
<br>&gt;<br>&gt; server is sending 404 as response for Subscribe, can anyone tell me it&#39;s<br>&gt; correct behaviour of server,<br>&gt; as that &lt;LIST&gt; is existing<br>&gt;<br>&gt; Thanks,<br>&gt; Vishal...<br>&gt; _______________________________________________ 
<br>&gt; Sip mailing list<br>&gt; <a onclick="return top.js.OpenExtLink(window,event,this)" href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank">https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt; This list is for NEW development of the core SIP Protocol
<br>&gt; Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip-implementors@cs.columbia.edu" target="_blank">sip-implementors@cs.columbia.edu</a> for questions on<br>&gt; current sip<br>&gt; Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sipping@ietf.org" target="_blank">
sipping@ietf.org</a> for new developments on the application of sip<br>&gt;<br></blockquote></div><br></span></div>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jul 20 03:59:42 2007
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Date: Fri, 20 Jul 2007 15:58:13 +0800
From: Qian Sun <sunqian@huawei.com>
Subject: RE: [Sip] comments on draft-camarillo-sip-body-handling-01
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To: 'Paul Kyzivat' <pkyzivat@cisco.com>,
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Hi,
I suggest Content Indirection Mechanism defined in rfc4483 should also be considered in this draft.

It is a little special, an example from rfc4483:
6.1.  Single Content Indirection

           INVITE sip:boromir@example.com SIP/2.0
           Accept: message/external-body application/sdp
           Content-Type: message/external-body;
                ACCESS-TYPE=URL;
                URL="http://www.example.net/party/06/2002/announcement";
                EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
                size=231
           Content-Length: 105

           Content-Type: application/sdp
           Content-Disposition: session
           Content-ID: <4e5562cd1214427d@example.net>

In addition, what happens to the INVITE request if the recipient can not get the indirect content?

Nits: a letter "l" is missing in the title.
Internet-Draft         Message Body Hand"l"ing in SIP              May 2007

Cheers,
Qian

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Friday, July 20, 2007 2:30 AM
To: Gonzalo Camarillo
Cc: IETF SIP List
Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01

Just one comment in addition to what Gonzalo said:

Gonzalo Camarillo wrote:

>> The use case described for multipart/alternative - new SDP mechanisms 
>> - seems kind of bogus. We've decided not to pursue that. Do we have a 
>> REAL use case for this?
> 
> AFAICT, we have decided not to pursue SDPng but we still need a 
> mechanism in order to be able to, at some point, migrate to new session 
> description formats. The current consensus, as I understand it, is that 
> multipart/alternative should not be used to provide alternative SDP 
> descriptions, but that it could be used to provide alternative session 
> descriptions written in different formats.

The other classic example, because it is just like email, is MESSAGE 
with a multipart/alternative containing plain text and html.

	Paul


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From sip-bounces@ietf.org Fri Jul 20 04:49:31 2007
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Date: Fri, 20 Jul 2007 09:49:18 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] RFC3891 (Replaces header) discrepancies
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In addition to what Dale has said, consider the case where call2 has
been forked to several destinations, 2a, 2b, 2c... To populate the
Replaces header field and the Refer-To URI, the sender of the REFER
request would need to pick one of these arbitrarily, say 2b. So suppose
the REFER request were sent and dereferenced, resulting in an INVITE
request with Replaces header field arriving at 2b. What if call2 is then
answered by 2a (or has already been answered by 2a while this signalling
is taking place)? I don't think the intended operation of attended
transfer would then be realised. So basically, I don't think the authors
of RFC 3891 saw any solutions to problems of this nature, and hence the
restriction.

John 

> -----Original Message-----
> From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net] 
> Sent: 19 July 2007 18:47
> To: sip@ietf.org
> Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies
> 
> 
>    From: "Jackson, James" <james_jackson@labs.att.com>
> 
>    Consider an attended transfer in which call1 is 
> established between a
>    transferee and a transferor. The transferor creates a new 
> call2 towards
>    transfer target. While this call is still alerting (early 
> dialog), it
>    sends a REFER on call1 with a Replaces header that 
> references call2. In
>    this case, the Replaces header attempts to replace an 
> early dialog which
>    was "terminated" by the target.
> 
> That is not allowed by RFC 3891.  But suppose it was allowed.  Further
> suppose that the transfer target rings for another 90 seconds (sending
> an additional 180) and then the proxy that manages the transfer target
> cancels the call, causing the transfer target to send a 487.  Who
> receives the 180 and 487?
> 
> Dale
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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From: Aki Niemi <aki.niemi@nokia.com>
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Subject: [Sip] Re: Model for entity-tags in subnot-etags
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Hi Dale,

ext sip-bounces@ietf.org wrote:
>    From: Aki Niemi <aki.niemi@nokia.com>
> 
>    - There is a single authoritative notifier agent responsible for any
>    single resource.
> 
>    - A resource is identified using a SIP URI.
> 
>    - A subscriber subscribes to a resource using the SIP URI.
> 
> Often a subscriber will send the SUBSCRIBE to a URI which is not the
> request-URI of the SUBSCRIBE when it reaches the notifier agent.  This
> happens due to normal SIP request routing, but the routing can also
> change with time.

Correct, but are you saying there might be more than one agent with a
different idea on the entity-tags of entities pertaining to a certain
resource? In other words, that the resource has multiple agents that may
have a different view of the resource state?

I believe this is disallowed per RFC3265 already.

>    - Each entity has an assigned entity-tag.
> 
> Each entity has a succession of versions.  Each version of each entity
> has an assigned entity-tag.

Yes.

>    - There is a one-to-many relationship between an entity and its versions
>    (different etag values, of which only one is current).
> 
>    - An entity-tag must be unique across all versions of all entities from
>    a resource.
> 
> Does this rule out hash-based entity-tags, which are only
> probabalistically unique?

No, I think this MUST statement should probably be followed with an
implementation note that talks about what's reasonably unique in this
context.

Cheers,
Aki


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From sip-bounces@ietf.org Fri Jul 20 06:04:55 2007
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Date: Fri, 20 Jul 2007 18:04:04 +0800
From: Qian Sun <sunqian@huawei.com>
Subject: RE: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides	for
	body (MIME) handling)
In-reply-to: <469FAA12.1030800@cisco.com>
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Hi,

If we define a new Content Disposition type "advertisement" so that the operator or service provider may insert advertisement in the all kinds of message body. And we want the advertisement content 'render' to the users. Could we use multiple content dispositions ("advertisement" and 'render') for a single content part? Or only define a new Content Disposition Parameters "advertisement" instead?

This is a real requirement, OMA and GSM is developing the mobile advertisement spec. the current advertisement way in Email or Webpage is very "evil", it is difficult for the user's client to distinguish the real content from advertisement, sometimes this is very disgusting or confusing. A new Content Disposition type "advertisement" or something else could be helpful, the advertisement may be displayed in a place different from the content.

Cheers,
Qian

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Friday, July 20, 2007 2:15 AM
To: Gonzalo Camarillo
Cc: sip
Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides for body (MIME) handling)



Gonzalo Camarillo wrote:
> Hi,
> 
> yes, I agree there is a bigger issue with 'render'. Even though this 
> bigger issue does not directly relate to multipart, we can also tackle 
> it in the same draft and clarify how 'render' is used and what it means.

Sounds good. Maybe its just a matter of changing the title of the draft 
to match the scope. Something like: 
draft-sipping-content-type-disp-clarification-nn.txt

	Paul

> Cheers,
> 
> Gonzalo
> 
> 
> Paul Kyzivat wrote:
>>
>>
>> Gonzalo Camarillo wrote:
>>> Hi Paul,
>>>
>>>> Regarding Content-Disposition in multiparts:
>>>>
>>>> While it seems "wrong" to me, I can live with using "render" as the 
>>>> normal disposition for multipart, *if* we can do a careful job of 
>>>> defining what "render" actually means for SIP. Used this way it 
>>>> clearly doesn't always mean to display the content to a user.
>>>
>>> An entity receiving a 'multipart/mixed' body will process its body 
>>> parts based on their disposition types.If the disposition type of a 
>>> particular body part is not understood, it will be displayed to the 
>>> user because the default disposition type is 'render'.
>>
>> There is a bigger problem with 'render' than just multipart.
>>
>> We have many uses of body parts that never mention Content-Disposition 
>> at all. The related examples don't use it, and so they all get a 
>> default of 'render'. This is particularly the case with events. E.g.
>>
>> - the notification for REFER contains type application/sipfrag.
>>   Its probably not intended to be "rendered" in any literal sense,
>>   though some conclusion about success or failure may be.
>>
>> - usage of filter bodies in presence subscriptions (RFC 4660)
>>   doesn't have a specified disposition, so it implicitly uses render.
>>   Its hard for me to convince myself that what the presence server
>>   does with these can be described as rendering.
>>
>> - In fact, the only place I can think of where 'render' is (implicitly)
>>   used in sip and has the appropriate connotation is in MESSAGE.
>>
>> I think we just have to grandfather the use of 'refer' in SIP, and 
>> document that the usage in sip does not mean "render", but rather 
>> means that the usage must be determined by other context - 
>> specifically the type of message or response and the content type. 
>> IOW, 'render' in sip can be thought of as a peculiar spelling for 
>> 'by-context'. It might be helpful to give guidance for development of 
>> future specs about when 'render' should be used vs defining a new 
>> disposition.
>>
>>> Therefore, the 'multipart/mixed' body will never be processed based 
>>> on its disposition type.
>>
>> Based on my discussion above, I would consider this a case of the 
>> multipart being processed according to its context *because* its 
>> content-disposition is 'refer'. If it had some other 
>> content-disposition then that might mean that it should be processed 
>> specially. E.g. if a multipart/mixed had disposition 'by-reference' 
>> then that needs to be taken into account, and the processing of the 
>> multipart becomes conditional on the reference.
>>
>> Also, there are only certain contexts in which multipart should be 
>> processed "normally". Consider a NOTIFY from a REFER, with a 
>> Content-Type of message/sipfrag, which happened to include a body as 
>> part of the fragment. (This is legal.) That body might contain a 
>> multipart/mixed, and that might contain other stuff. But I think the 
>> processing of all of that is dependent on processing of the sipfrag. 
>> For instance, if something inside that had handling=required, I don't 
>> think that means that processing of the NOTIFY fails because the 
>> recipient can't handle that.
>>
>>> This means that it does not make much sense to talk about its 
>>> disposition type as a whole.
>>
>> Counterexample above.
>>
>>> However, we need to have one in order to be able to set its handling 
>>> parameter. That's why we can either keep 'render' explaining that it 
>>> is actually never used or define a new disposition type.
>>
>> I can go either way on that. But because there is existing usage, and 
>> because 'render' semantics are already so messed up, I think it makes 
>> more sense to stick with 'render' and just clarify its meaning.
>>
>>> Since existing implementations will not understand our newly defined 
>>> disposition type, they will treat it as 'render' anyway... but the 
>>> resulting behavior would be the same. That's why it may make sense to 
>>> keep using 'render'.
>>
>> Yup.
>>
>>> On the other hand, semantically, it would make more sense to define a 
>>> new disposition type.
>>>
>>> Also, if we decide to define the 'by-reference' disposition type (per 
>>> a different email thread), it could be used here as well.
>>
>> It could be used when there is actually a reference to it. But it 
>> would not be an appropriate disposition to use when there is no 
>> reference to the multipart.
>>
>>     Thanks,
>>     Paul
>>
>>> Cheers,
>>>
>>> Gonzalo
>>>
>>>
>>>
>>>> Gonzalo Camarillo wrote:
>>>>> Folks,
>>>>>
>>>>> here you have the slides I have put together for the body (MIME) 
>>>>> handling presentation on Monday:
>>>>>
>>>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>>>
>>>>>
>>>>> These slides relate to this draft:
>>>>>
>>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>>>
>>>>>
>>>>> The slides discuss all the open issues that have been brought up in 
>>>>> the list in the last months.
>>>>>
>>>>> Comments are welcome.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Gonzalo
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>>
>>>
> 


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Qian Sun wrote:
> If we define a new Content Disposition type "advertisement" so that
> the operator or service provider may insert advertisement in the all
> kinds of message body. And we want the advertisement content 'render'
> to the users. Could we use multiple content dispositions
> ("advertisement" and 'render') for a single content part? Or only
> define a new Content Disposition Parameters "advertisement" instead?

So is there a need to ever use "advertisement" in conjunction with 
something besides "render"?

--
Dean



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   From: Aki Niemi <aki.niemi@nokia.com>

   ext sip-bounces@ietf.org wrote:
   >    From: Aki Niemi <aki.niemi@nokia.com>
   > 
   >    - There is a single authoritative notifier agent responsible for any
   >    single resource.
   > 
   >    - A resource is identified using a SIP URI.
   > 
   >    - A subscriber subscribes to a resource using the SIP URI.
   > 
   > Often a subscriber will send the SUBSCRIBE to a URI which is not the
   > request-URI of the SUBSCRIBE when it reaches the notifier agent.  This
   > happens due to normal SIP request routing, but the routing can also
   > change with time.

   Correct, but are you saying there might be more than one agent with a
   different idea on the entity-tags of entities pertaining to a certain
   resource? In other words, that the resource has multiple agents that may
   have a different view of the resource state?

   I believe this is disallowed per RFC3265 already.

I doubt things could work with multiple agents that have a different
view of the resource state.

But there might be multiple agents with different addresses, from whom
you request the state of the same resource using different SIP URIs.
(Which ultimately might all be destinations reached by sending
requests to the same AOR.)

I don't think it interferes with the subnot mechanism, but my point is
that a resource might *not* have a single, specific SIP URI, either as
the request-URI when the SUBSCRIBE reaches the agent(s) which can
deliver its state, or as the address to which UAs subscribe.

All you need to make subnot work is that every UA agrees on which
subscriptions are "for the same resource", and that all notifiers for
"the same resource" generate the same etags for the same state, and
different etags for different state.

Dale


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Qian Sun wrote:
> Hi,
> 
> If we define a new Content Disposition type "advertisement" so that the operator or service provider may insert advertisement in the all kinds of message body. And we want the advertisement content 'render' to the users. Could we use multiple content dispositions ("advertisement" and 'render') for a single content part? Or only define a new Content Disposition Parameters "advertisement" instead?

I'm quite certain that there may be only one content disposition per 
body part, just as there can be only one content type per body part.

> This is a real requirement, OMA and GSM is developing the mobile advertisement spec. the current advertisement way in Email or Webpage is very "evil", it is difficult for the user's client to distinguish the real content from advertisement, sometimes this is very disgusting or confusing. A new Content Disposition type "advertisement" or something else could be helpful, the advertisement may be displayed in a place different from the content.

I don't think I want to think or talk about "advertisement".

	Paul

> Cheers,
> Qian
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Friday, July 20, 2007 2:15 AM
> To: Gonzalo Camarillo
> Cc: sip
> Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides for body (MIME) handling)
> 
> 
> 
> Gonzalo Camarillo wrote:
>> Hi,
>>
>> yes, I agree there is a bigger issue with 'render'. Even though this 
>> bigger issue does not directly relate to multipart, we can also tackle 
>> it in the same draft and clarify how 'render' is used and what it means.
> 
> Sounds good. Maybe its just a matter of changing the title of the draft 
> to match the scope. Something like: 
> draft-sipping-content-type-disp-clarification-nn.txt
> 
> 	Paul
> 
>> Cheers,
>>
>> Gonzalo
>>
>>
>> Paul Kyzivat wrote:
>>>
>>> Gonzalo Camarillo wrote:
>>>> Hi Paul,
>>>>
>>>>> Regarding Content-Disposition in multiparts:
>>>>>
>>>>> While it seems "wrong" to me, I can live with using "render" as the 
>>>>> normal disposition for multipart, *if* we can do a careful job of 
>>>>> defining what "render" actually means for SIP. Used this way it 
>>>>> clearly doesn't always mean to display the content to a user.
>>>> An entity receiving a 'multipart/mixed' body will process its body 
>>>> parts based on their disposition types.If the disposition type of a 
>>>> particular body part is not understood, it will be displayed to the 
>>>> user because the default disposition type is 'render'.
>>> There is a bigger problem with 'render' than just multipart.
>>>
>>> We have many uses of body parts that never mention Content-Disposition 
>>> at all. The related examples don't use it, and so they all get a 
>>> default of 'render'. This is particularly the case with events. E.g.
>>>
>>> - the notification for REFER contains type application/sipfrag.
>>>   Its probably not intended to be "rendered" in any literal sense,
>>>   though some conclusion about success or failure may be.
>>>
>>> - usage of filter bodies in presence subscriptions (RFC 4660)
>>>   doesn't have a specified disposition, so it implicitly uses render.
>>>   Its hard for me to convince myself that what the presence server
>>>   does with these can be described as rendering.
>>>
>>> - In fact, the only place I can think of where 'render' is (implicitly)
>>>   used in sip and has the appropriate connotation is in MESSAGE.
>>>
>>> I think we just have to grandfather the use of 'refer' in SIP, and 
>>> document that the usage in sip does not mean "render", but rather 
>>> means that the usage must be determined by other context - 
>>> specifically the type of message or response and the content type. 
>>> IOW, 'render' in sip can be thought of as a peculiar spelling for 
>>> 'by-context'. It might be helpful to give guidance for development of 
>>> future specs about when 'render' should be used vs defining a new 
>>> disposition.
>>>
>>>> Therefore, the 'multipart/mixed' body will never be processed based 
>>>> on its disposition type.
>>> Based on my discussion above, I would consider this a case of the 
>>> multipart being processed according to its context *because* its 
>>> content-disposition is 'refer'. If it had some other 
>>> content-disposition then that might mean that it should be processed 
>>> specially. E.g. if a multipart/mixed had disposition 'by-reference' 
>>> then that needs to be taken into account, and the processing of the 
>>> multipart becomes conditional on the reference.
>>>
>>> Also, there are only certain contexts in which multipart should be 
>>> processed "normally". Consider a NOTIFY from a REFER, with a 
>>> Content-Type of message/sipfrag, which happened to include a body as 
>>> part of the fragment. (This is legal.) That body might contain a 
>>> multipart/mixed, and that might contain other stuff. But I think the 
>>> processing of all of that is dependent on processing of the sipfrag. 
>>> For instance, if something inside that had handling=required, I don't 
>>> think that means that processing of the NOTIFY fails because the 
>>> recipient can't handle that.
>>>
>>>> This means that it does not make much sense to talk about its 
>>>> disposition type as a whole.
>>> Counterexample above.
>>>
>>>> However, we need to have one in order to be able to set its handling 
>>>> parameter. That's why we can either keep 'render' explaining that it 
>>>> is actually never used or define a new disposition type.
>>> I can go either way on that. But because there is existing usage, and 
>>> because 'render' semantics are already so messed up, I think it makes 
>>> more sense to stick with 'render' and just clarify its meaning.
>>>
>>>> Since existing implementations will not understand our newly defined 
>>>> disposition type, they will treat it as 'render' anyway... but the 
>>>> resulting behavior would be the same. That's why it may make sense to 
>>>> keep using 'render'.
>>> Yup.
>>>
>>>> On the other hand, semantically, it would make more sense to define a 
>>>> new disposition type.
>>>>
>>>> Also, if we decide to define the 'by-reference' disposition type (per 
>>>> a different email thread), it could be used here as well.
>>> It could be used when there is actually a reference to it. But it 
>>> would not be an appropriate disposition to use when there is no 
>>> reference to the multipart.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>>> Cheers,
>>>>
>>>> Gonzalo
>>>>
>>>>
>>>>
>>>>> Gonzalo Camarillo wrote:
>>>>>> Folks,
>>>>>>
>>>>>> here you have the slides I have put together for the body (MIME) 
>>>>>> handling presentation on Monday:
>>>>>>
>>>>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.ppt 
>>>>>>
>>>>>>
>>>>>> These slides relate to this draft:
>>>>>>
>>>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.txt 
>>>>>>
>>>>>>
>>>>>> The slides discuss all the open issues that have been brought up in 
>>>>>> the list in the last months.
>>>>>>
>>>>>> Comments are welcome.
>>>>>>
>>>>>> Thanks,
>>>>>>
>>>>>> Gonzalo
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol
>>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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From sip-bounces@ietf.org Fri Jul 20 09:15:32 2007
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This is to give an overview of the current work in the SIP WG

Regards

Keith

Documents published since IETF#68 =3D 2
-------------------------------------

draft-ietf-sip-mib-12 as RFC 4780 (Proposed standard) (Thanks to Kevin
Lingle, Jean-Francois Mule, Joon Maeng, Dave Walker)

draft-ietf-sip-connected-identity-05  (Proposed standard) as RFC 4916
(Thanks to John Elwell)

Documents in RFC editor's queue =3D none
--------------------------------------

Documents now with IESG =3D 11
----------------------------

draft-ietf-sip-e2m-sec-06 (Proposed standard)
Dean Willis is document shepherd=20
IESG last call completed 14th May 2007. No last call comments. Problems
with proposed standard status
Current status: Waiting for AD Go ahead

draft-ietf-sip-fork-loop-fix-05 (Proposed standard)
Keith Drage is document shepherd
IESG last call completed 14th May 2007.=20
No last call comments. Security directorate review identified one
comment, and one open comment exists on SIP list. Will be discussing at
forthcoming meeting
Current status: Waiting for AD Go ahead:: Revised ID needed

draft-ietf-sip-acr-code-05 (Proposed standard)
Dean Willis is document shepherd
IESG last call initiated 6th July 2007 to complete 19th July 2007.
Currently in state: Approved-announcement to be sent :: Point Raised -
writeup needed.

draft-ietf-sip-answermode-02 (Proposed standard)
Keith Drage is document shepherd
IESG last call initiated 10th July 2007 to complete 5th August 2007.
Currently in state: In last call.

draft-ietf-sip-gruu-14 (Proposed standard)
Keith Drage is document shepherd
IESG last call initiated 2nd July 2007 to complete 16th July 2007
Currently in state: IESG evaluation.

draft-ietf-sip-multiple-refer-01  (Proposed standard)
Keith Drage is document shepherd
Currently in publication requested - external party, waiting on
publication request for draft-ietf-sip-consent-framework

draft-ietf-sip-uri-list-message-01  (Proposed standard)
Keith Drage is document shepherd
Currently in publication requested - external party, waiting on
publication request for draft-ietf-sip-consent-framework

draft-ietf-sip-uri-list-conferencing-01  (Proposed standard)
Keith Drage is document shepherd
Currently in publication requested - external party, waiting on
publication request for draft-ietf-sip-consent-framework

draft-ietf-sip-uri-list-subscribe-01  (Proposed standard)
Keith Drage is document shepherd
Currently in publication requested - external party, waiting on
publication request for draft-ietf-sip-consent-framework

draft-ietf-sip-consent-framework-02 (Proposed standard)
Keith Drage is document shepherd
Publication requested 13th July 2007
Currently in state: Publication requested

draft-ietf-sip-ice-option-tag-02 (Proposed standard)
Keith Drage is PROTO document shepherd
Publication requested 17th July 2007
Currently in state: AD evaluation

Documents completed last call awaiting submission to IESG =3D 1
-------------------------------------------------------------

draft-ietf-sip-certs-04 (Proposed standard)
Dean Willis is document shepherd
Milestone for sending to IESG is June 2007
Token currently with author: waiting on one open issue before new
version and submission

Documents in WGLC =3D 5
---------------------

draft-ietf-sip-connect-reuse-07
Candidate: Proposed standard
Milestone to IESG August 2007
WGLC began 26th October 2005
Has some form of dependency on=20
draft-gurbani-sip-domain-certs-03=20
Alternate proposal: draft-rosenberg-sip-outbound-discovery-mid-dialog-00
Currently WG chairs and authors evaluating what we need in the roadmap
here
Direction forward?

draft-ietf-sip-location-conveyance-08=20
Candidate: Proposed standard
Milestone for submission to IESG is July 2007
Review team assembled: Authors, WG chairs, Shida Schubert, Joszef Varga,
Henning Schulzrinne, Richard Barnes, Hannes Tschofenig
WGLC initiated 4th March 2007 to complete 2nd April 2007. WGLC also
issued in GEOPRIV WG
See
http://www.softarmor.com/sipwg/reviews/location-conveyance/index.html
for current status of review. New version (-08) issued but still working
WGLC comments
Keith Drage is PROTO document shepherd

draft-ietf-sip-session-policy-framework-01=20
Candidate: Proposed standard
SIPPING WG review initiated 4th August 2006 to complete 25th August 2006
Transferred from SIPPING group because contains elements RFC 3427 says
must be done by SIP
Related documents in SIPPING:
draft-ietf-sipping-policy-package-03=20
draft-ietf-sipping-media-policy-dataset-03
No major WGLC comments - document to be submitted when revised draft
available
Keith Drage is PROTO document shepherd
Milestone for submission to IESG September 2007

draft-ietf-sip-sips-05=20
Candidate: Proposed standard
Milestone for WGLC June 2007
WGLC initiated 29th June 2007 to complete 16th July 2007
Dean Willis is PROTO document shepherd
Milestone for submission to IESG September 2007

draft-ietf-sip-outbound-10
Candidate: Proposed standard
Milestone for WGLC is March 2007
WGLC initiated 9th July 2007 to complete 6th August 2007
Milestone for sending to IESG is June 2007
Review team to supplement WGLC. Review team so far: Authors, WG chairs,
Jonathan Rosenberg, Fredrick Thulin, Kevin Johns, Christer Holmberg, AC
Mahendran, Alan Hawrylyshen, Byron Campen, Derek McDonald, EKR, Nils
Ohlmeier, Francois Audet
Dean Willis is PROTO document shepherd

WG documents still being developed =3D 8
--------------------------------------

draft-ietf-sip-hop-limit-diagnostics-03
Candidate: Proposed standard
Milestone for WGLC is July 2007
Milestone for submission to IESG is November 2007
Scope of charter item is more general than hop limit diagnostics
Key issue is size of response when it has to traverse UDP transports.
Editor has procedures for reducing size of response body - does this
render the mechanism of limited use if this occurs
Issue has resulted in generation of:
draft-gurbani-sip-large-udp-response-00=20
draft-petithuguenin-sip-fragmentation-responses-02
Should document go forward as is?
Draft expired and removed from drafts directory - where next - WG polled


draft-ietf-sip-sec-flows-01
Candidate: Informational
Milestone for WGLC is July 2007
Milestone for submission to IESG is September 2007
To a certain extent dependent on resolution of ideas in (and therefore
waiting on them):
draft-ietf-sip-sips-02 =20
draft-gurbani-sip-domain-certs-04=20
Draft expired and removed from drafts directory=20

draft-ietf-sip-saml-02
Candidate: Proposed standard
Milestone for WGLC is September 2007
Milestone for submission to IESG is December 2007

draft-ietf-sip-hitchhikers-guide-03
Candidate: Informational
Milestone for WGLC December 2007
Milestone for submission to IESG December 2007
See also: draft-sinnreich-sip-tools-01

draft-ietf-sip-xcap-config-00=20
Candidate: proposed standard
Milestone for WGLC July 2007
Milestone for submission to IESG December 2007
Document transferred from SIPPING because of potential need for new
response code
Related documents in SIPPING
draft-ietf-sipping-config-framework-11
draft-petrie-sipping-identity-dataset-01=20
draft-petrie-sipping-profile-datasets-04=20
draft-petrie-sipping-sip-dataset-02=20
draft-petrie-sipping-voip-features-dataset-01=20
See also draft-petrie-sip-event-param-err-00.txt
Waiting on SIPPING rewrite of draft-ietf-sipping-config-framework
OMA liaison - possible interaction with draft-ietf-simple-xcap-diff and
draft-urpalainen-sip-xcap-diff-event-01

draft-rosenberg-sip-ua-loose-route-00=20
Agreed as WG item (subject to AD) at IETF#67
Candidate: proposed standard
December 2007: Delivering request-URI and parameters to UAS via proxy to
WGLC
February 2008: Delivering request-URI and parameters to UAS via proxy to
IESG (PS)
Author to submit sip-00 version

draft-ietf-sip-subnot-etags-00
Agreed as WG item (subject to AD) at IETF#67
Candidate: proposed standard
September 2007 -- Extension for use of etags in conditional notification
to WGLC
December 2007 -- Extension for use in etags in conditional notification
to IESG (PS)

draft-ietf-sip-record-route-fix-00
Essential correction?



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There are some existing SIP server implementations that support REFER/ 
Replaces in the situation that James describes. It only works,  
though, if you *know* that there is exactly one dialog involved in  
the call -- i.e. if you *know* that no forking has occurred.

Sending REFER/Replaces to complete a transfer before the transfer  
target has answered the call is understandably seen as desirable,  
because it allows the transferor to entirely remove itself from the  
call at the time of the transfer. But for the reasons John and Dale  
have noted, it's not as simple as that. The safe, RFC 3892-compliant  
way to handle this transfer is for the transferor to continue  
managing the two dialogs (transferee and transfer target) until the  
transfer target dialog has become confirmed, at which time the REFER  
is performed. The transferor UA can do this "in the background" --  
the device is immediately available to the user for initiating or  
receiving a new call, even though the UA is waiting for the transfer  
target to answer.

I've had to explain this a number of times to people who don't  
recognize how forking makes SIP different from traditional switched  
telephone signaling. They tend to build things without recognizing  
the possibility of forking.

On Jul 20, 2007, at 4:49 AM, Elwell, John wrote:

> In addition to what Dale has said, consider the case where call2 has
> been forked to several destinations, 2a, 2b, 2c... To populate the
> Replaces header field and the Refer-To URI, the sender of the REFER
> request would need to pick one of these arbitrarily, say 2b. So  
> suppose
> the REFER request were sent and dereferenced, resulting in an INVITE
> request with Replaces header field arriving at 2b. What if call2 is  
> then
> answered by 2a (or has already been answered by 2a while this  
> signalling
> is taking place)? I don't think the intended operation of attended
> transfer would then be realised. So basically, I don't think the  
> authors
> of RFC 3891 saw any solutions to problems of this nature, and hence  
> the
> restriction.
>
> John
>
>> -----Original Message-----
>> From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]
>> Sent: 19 July 2007 18:47
>> To: sip@ietf.org
>> Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies
>>
>>
>>    From: "Jackson, James" <james_jackson@labs.att.com>
>>
>>    Consider an attended transfer in which call1 is
>> established between a
>>    transferee and a transferor. The transferor creates a new
>> call2 towards
>>    transfer target. While this call is still alerting (early
>> dialog), it
>>    sends a REFER on call1 with a Replaces header that
>> references call2. In
>>    this case, the Replaces header attempts to replace an
>> early dialog which
>>    was "terminated" by the target.
>>
>> That is not allowed by RFC 3891.  But suppose it was allowed.   
>> Further
>> suppose that the transfer target rings for another 90 seconds  
>> (sending
>> an additional 180) and then the proxy that manages the transfer  
>> target
>> cancels the call, causing the transfer target to send a 487.  Who
>> receives the 180 and 487?
>>
>> Dale
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jul 20 10:36:35 2007
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From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <200707191003.l6JA3TVe010387@imf.m.ecl.ntt.co.jp>
To: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>, sip@ietf.org
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Mayumi,

I have looked through the privacy-guidelines document and have the
following comments.

1. I was not aware of TISPAN use of priv-values other than "id". This
does indeed give greater justification for this draft, but I would
recommend you seek review from a couple of TISPAN experts.

2. Should this in fact be targeted as a BCP?

3. There seems to be no mention of impact on event packages whose
notifications might convey information that has been anonymized or
removed. I am thinking in particular about Call-Id and the Dialog event
package, where an anonymizer would need to intervene. Of course, in this
particular case, problems can be avoided by not authorizing subscription
- I don't know whether this is a good enough solution.

4. The Join header field is subject to the same issues as Replaces and
Target-Dialog.

Otherwise it looks to be in good shape.

John
 

> -----Original Message-----
> From: Mayumi Munakata [mailto:munakata.mayumi@lab.ntt.co.jp] 
> Sent: 19 July 2007 11:03
> To: Elwell, John
> Cc: shida@ntt-at.com; ohba.takumi@lab.ntt.co.jp
> Subject: Re: draft-munakata-sip-privacy-new/guideline
> 
> John,
> 
> Thank you for the comments for the privacy drafts.
> See inline.
> 
> > [JRE] I guess the question we have to ask is whether there are
> > substantical interoperability problems with RFC 3323. As 
> the only value
> > that seems to be used in practice is 'id', rules are fairly
> > straightforward.
> 
> TISPAN specs are actually using other priv-values such as 
> 'user' and 'header'.  An interoperability problem could 
> occur because of different interpretations of priv-values.
> 
> > One concern, however, is with 'history'. I don't know 
> whether this is
> > implemented in practice. RFC 4244 allows a proxy to insert 
> this value.
> > So whilst privacy-new defines a new way for UAs to ensure 
> privacy and
> > should in the longer term lead to a move away from the 
> Privacy header
> > field (depending on what is used for the privacy flag), 
> what will happen
> > to proxy use of Privacy: history?
> 
> The privacy flag to be defined in the privacy-new draft 
> will be a request for proxies to delete History-Info headers 
> on the edge of Trust Domain.
> However, as you say, 'history' privacy is usually requested
> by a proxy for the purpose of topology hiding.
> The "privacy" in this new mechanism means the "user-privacy" 
> so the "network-privacy" (topology hiding) may be out of scope.
> 
> Thanks,
> Mayumi
> 
> 


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Subject: RE: [Sip] RFC3891 (Replaces header) discrepancies
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From: "Jackson, James" <james_jackson@labs.att.com>
To: "David Robbins" <robbins.dave@verizon.net>,
	"Elwell, John" <john.elwell@siemens.com>
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Thanks to everyone for the clarification on this !

James

-----Original Message-----
From: David Robbins [mailto:robbins.dave@verizon.net]=20
Sent: Friday, July 20, 2007 8:23 AM
To: Elwell, John
Cc: sip@ietf.org
Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies

There are some existing SIP server implementations that support REFER/=20
Replaces in the situation that James describes. It only works, =20
though, if you *know* that there is exactly one dialog involved in =20
the call -- i.e. if you *know* that no forking has occurred.

Sending REFER/Replaces to complete a transfer before the transfer =20
target has answered the call is understandably seen as desirable, =20
because it allows the transferor to entirely remove itself from the =20
call at the time of the transfer. But for the reasons John and Dale =20
have noted, it's not as simple as that. The safe, RFC 3892-compliant =20
way to handle this transfer is for the transferor to continue =20
managing the two dialogs (transferee and transfer target) until the =20
transfer target dialog has become confirmed, at which time the REFER =20
is performed. The transferor UA can do this "in the background" -- =20
the device is immediately available to the user for initiating or =20
receiving a new call, even though the UA is waiting for the transfer =20
target to answer.

I've had to explain this a number of times to people who don't =20
recognize how forking makes SIP different from traditional switched =20
telephone signaling. They tend to build things without recognizing =20
the possibility of forking.

On Jul 20, 2007, at 4:49 AM, Elwell, John wrote:

> In addition to what Dale has said, consider the case where call2 has
> been forked to several destinations, 2a, 2b, 2c... To populate the
> Replaces header field and the Refer-To URI, the sender of the REFER
> request would need to pick one of these arbitrarily, say 2b. So =20
> suppose
> the REFER request were sent and dereferenced, resulting in an INVITE
> request with Replaces header field arriving at 2b. What if call2 is =20
> then
> answered by 2a (or has already been answered by 2a while this =20
> signalling
> is taking place)? I don't think the intended operation of attended
> transfer would then be realised. So basically, I don't think the =20
> authors
> of RFC 3891 saw any solutions to problems of this nature, and hence =20
> the
> restriction.
>
> John
>
>> -----Original Message-----
>> From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]
>> Sent: 19 July 2007 18:47
>> To: sip@ietf.org
>> Subject: Re: [Sip] RFC3891 (Replaces header) discrepancies
>>
>>
>>    From: "Jackson, James" <james_jackson@labs.att.com>
>>
>>    Consider an attended transfer in which call1 is
>> established between a
>>    transferee and a transferor. The transferor creates a new
>> call2 towards
>>    transfer target. While this call is still alerting (early
>> dialog), it
>>    sends a REFER on call1 with a Replaces header that
>> references call2. In
>>    this case, the Replaces header attempts to replace an
>> early dialog which
>>    was "terminated" by the target.
>>
>> That is not allowed by RFC 3891.  But suppose it was allowed.  =20
>> Further
>> suppose that the transfer target rings for another 90 seconds =20
>> (sending
>> an additional 180) and then the proxy that manages the transfer =20
>> target
>> cancels the call, causing the transfer target to send a 487.  Who
>> receives the 180 and 487?
>>
>> Dale
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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   From: David Robbins <robbins.dave@verizon.net>

Thanks for the explanation!

   The safe, RFC 3892-compliant way to handle this transfer is for the
   transferor to continue managing the two dialogs (transferee and
   transfer target) until the transfer target dialog has become
   confirmed, at which time the REFER is performed. The transferor UA
   can do this "in the background" -- the device is immediately
   available to the user for initiating or receiving a new call, even
   though the UA is waiting for the transfer target to answer.

I know the (now obsolete) Xpressa phone from Pingtel does that.  I
expect many modern phones do so also.

Dale


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At this point in time, I think yes.

RjS

On Jul 20, 2007, at 8:15 AM, DRAGE, Keith ((Keith)) wrote:

> draft-ietf-sip-record-route-fix-00
> Essential correction?


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<HTML><BODY style="word-wrap: break-word; -khtml-nbsp-mode: space; -khtml-line-break: after-white-space; "><DIV>At this point in time, I think yes.</DIV><DIV><BR class="khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR><DIV><DIV>On Jul 20, 2007, at 8:15 AM, DRAGE, Keith ((Keith)) wrote:</DIV><BR class="Apple-interchange-newline"><BLOCKQUOTE type="cite"><P style="margin: 0.0px 0.0px 0.0px 0.0px"><FONT face="Helvetica" size="3" style="font: 12.0px Helvetica">draft-ietf-sip-record-route-fix-00</FONT></P> <P style="margin: 0.0px 0.0px 0.0px 0.0px"><FONT face="Helvetica" size="3" style="font: 12.0px Helvetica">Essential correction?</FONT></P> </BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>
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On Jul 19, 2007, at 5:04 PM, Jeroen van Bemmel wrote:

> Hi,
>
> A comment / clarification on outbound-10 section 5.3 Forwarding  
> requests, regarding flow token construction and incoming/outgoing  
> logic:
> The text says to send a 430 (Flow Failed) "For connection-oriented  
> transports, if the flow no longer exists" (SHOULD)
>
> This can be read as saying that if the original TCP connection no  
> longer exists, a 430 should be sent. However, if the UAC somehow  
> lost its connection and recovered, there would be a new TCP  
> connection with (likely) a different ephemeral port at the UAC.

There is a timing issue here.  Two things could happen based on timing.

1) If the UA hasn't noticed that its original TCP connection failed,  
or it noticed but the REGISTER for its new flow has not been received  
or processed yet, then edge proxy (if any) should send a 430.  If  
there is another active flow, the authoritative proxy can try that  
flow. The failed flow does not

2) The UA sent a new REGISTER to replace its original (failed) flow  
and that request has been processed.  The original flow would never  
be attempted.

In both cases, the best possible thing that could happen will happen.


> For the proposed flow token algorithm, this would cause the  
> described incoming/outgoing logic to fail (since the token no  
> longer matches an incoming flow from the UAC). Moreover, for  
> requests in the other direction ("incoming") the proxy would not be  
> able to locate the new flow, and send a 430 where it could have  
> done better.
>
> Two thoughts:
> 1. To determine if a (non-REGISTER) request is incoming/outgoing,  
> it might be easier to simply say: 1 Via header --> outgoing, else  
> incoming.

In a proxy with a colocated UA (ex: the registration package or a  
presence server), the proxy can emit "incoming" requests with 1 Via  
header.

> There are other options, the Edge proxy could check the request URI  
> for a Contact URI it has seen in REGISTER before. It could also be  
> based on Service-Route and/or Path.

This would require the Edge proxy to mirror registration state. We  
had a requirement that outbound should work without requiring this  
state mirroring.

> Perhaps this draft need not explain all details, but it currently  
> only describes logic for mid-dialog requests, and appears to fail  
> in the above described case of TCP connection loss.

What you are describing is a mid-dialog flow switchover.  This was so  
controversial that we made this explicitly out of the scope of this  
document.  The document does not prevent an edge proxy from doing  
this, but it doesn't describe how to do it either.

> 2. An alternative implementation could be to maintain a mapping of  
> Contact address ( IP/host+port ) to flow, which is updated each  
> time a REGISTER 200 OK comes by. If the UAC is using the same  
> Contact URIs as in its REGISTER (or a GRUU rewritten by the home  
> proxy), there is no need for a flow token at all: the Edge proxy  
> can use (IP/host+port part of) the request URI as an (hash) index  
> into the flow table (for incoming requests).

I'm not understanding where you propose to maintain this mapping.  If  
you want the edge proxy to keep per-flow state, I'm afraid we had a  
strong consensus not to add mechanism that requires that approach.    
If the UAC uses a GRUU in the Contact header, the edge proxy may not  
need to Record-Route.  However, that is a topic for either the GRUU  
spec or another spec on mid-dialog Record-Routing to cover.

> There are many other designs possible. Main point is that the  
> stateless flow token approach described seems suboptimal in case of  
> TCP connection loss (home proxy may be able to recover if the UAC  
> used 2 or more flows, but if all were lost and the backup edge  
> proxy also implemented this algorithm, ongoing dialogs would be lost)

Yes.  And we decided for this document that losing those calls was  
acceptable for this document.  Other specification work can take up  
that challenge, otherwise we will never make forward progress.

thanks,
-rohan

> Regards,
> Jeroen



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From sip-bounces@ietf.org Fri Jul 20 11:53:45 2007
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On 7/19/07 7:15 PM, Jeroen van Bemmel wrote:
> Good points, some nits inline

Good nits. Some responses inline.

>>
>> Why are we doing this so very, very wrong? It's almost as if we've
>> gone out of our way to violate both the spirit AND the law of feature
>> tag handling in as many ways as possible. Most annoyingly, we get
>> exactly the desired behavior if we just do things the way they're 
>> described in
>> 3261. To wit:
>>
>>    * If a client supports outbound, it includes "Supported: outbound"
>>      in all REGISTER requests, regardless of whether the specific
>>      request makes use of that extension.
>>
>
> [jvb] this would probably be a MUST, right?

Well, we're not strengthening 3261 -- so I believe it's a SHOULD in all 
requests, and a MUST in REGISTERs that make use of the extension.


>>  
>>    * The registrar includes "Require: outbound" in any REGISTER
>>      requests to which outbound behavior is being applied.
>
>
> [jvb] you mean responses

Right.

>>    * The registrar is free to include "Supported: outbound" in any
>>      responses it generates -- and it means exactly what 3261 intends
>>      it to mean: that outbound is *supported*, but is not being
>>      applied to the response in question.
>
>
> [jvb] is not _necessarily_ being applied, it depends on presence of 
> Require: outbound (only). Guess this is a MAY, right?

I believe you are correct on both counts.

/a


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From sip-bounces@ietf.org Fri Jul 20 12:06:34 2007
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Hi Adam,

What can I say but "You're right".  Unless there is some fundamental  
objection, I will make this change pronto to my working copy.

I believe this also means I can remove the 'ob' parameter from the  
UAC's Contact header.
thanks,
-rohan


On Jul 19, 2007, at 5:05 PM, Adam Roach wrote:
> At the risk of further bloodying the corpse of this poor horse, I  
> still think the usage of feature tags in outbound is way, way off  
> in terms of the semantics described in 3261.
>
> Reviewing the way this is defined to work in SIP:
>
> 3261:8.1.1.9:
>>    If the UAC supports extensions to SIP that can be applied by the
>>    server to the response, the UAC SHOULD include a Supported header
>>    field in the request listing the option tags (Section 19.2) for  
>> those
>>    extensions.
>
> 3261:8.2.4:
>
>>    A UAS that wishes to apply some extension when generating the
>>    response MUST NOT do so unless support for that extension is
>>    indicated in the Supported header field in the request.  If the
>>    desired extension is not supported, the server SHOULD rely only on
>>    baseline SIP and any other extensions supported by the client.
>> ...
>>    Any extensions applied to a non-421 response MUST be listed in a
>>    Require header field included in the response.  Of course, the  
>> server
>>    MUST NOT apply extensions not listed in the Supported header  
>> field in
>>    the request.  As a result of this, the Require header field in a
>>    response will only ever contain option tags defined in standards-
>>    track RFCs.
>
> There are five normative statements in the text I quote above. The  
> mechanism described in the outbound draft actually manages to  
> violate _all_ _five_ of them.
>
> The registrar infers *both* support for outbound *and* a request to  
> apply the outbound extension by the presence of the "reg-id"  
> Contact header field parameter. In particular, the client is  
> apparently NOT supposed to send a "Supported: outbound" header  
> field. This appears to contravene the normative statement in  
> 8.1.1.9 cited above.
>
> The registrar indicates that outbound procedures are being applied  
> by including a "Supported" header field in the REGISTER response.  
> (In 3261, "Supported" in a _response_ indicates that an extension  
> _can_ _be_ supported in a future request, NOT that it is being  
> applied to the current one -- sections 11 and 13.3.1.4 both talk  
> about Supported usage in responses.)
>
> In other words, the registrar (acting as a UAS) is applying an  
> extension when generating a response without that extension being  
> indicated in the "Supported" header field in the request. Again,  
> this directly contravenes two normative statements from section  
> 8.2.4 (cited above).
>
> The registrar is further applying an extension to a non-421  
> response that is not listed in a "Require" header field, in  
> violation of the remaining normative statement in the portion of  
> section 8.2.4 cited above.
>
> Why are we doing this so very, very wrong? It's almost as if we've  
> gone out of our way to violate both the spirit AND the law of  
> feature tag handling in as many ways as possible. Most annoyingly,  
> we get exactly the desired behavior if we just do things the way  
> they're described in 3261. To wit:
>
>    * If a client supports outbound, it includes "Supported: outbound"
>      in all REGISTER requests, regardless of whether the specific
>      request makes use of that extension.
>
>    * The client adds a "reg-id" header field parameter to any Contacts
>      to which outbound processing is to apply.
>
>    * The registrar includes "Require: outbound" in any REGISTER
>      requests to which outbound behavior is being applied.
>
>    * The registrar is free to include "Supported: outbound" in any
>      responses it generates -- and it means exactly what 3261 intends
>      it to mean: that outbound is *supported*, but is not being  
> applied
>      to the response in question.
>
> /a



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Date: Fri, 20 Jul 2007 09:26:44 -0700
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Subject: [Sip] Re: Outbound: Feature Tag (Ab)Use
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On Jul 20, 2007, at 9:06 AM, Rohan Mahy wrote:

> Hi Adam,
>
> What can I say but "You're right".  Unless there is some  
> fundamental objection, I will make this change pronto to my working  
> copy.
>
> I believe this also means I can remove the 'ob' parameter from the  
> UAC's Contact header.

On further thought, I realized that the 'ob' parameter is still  
needed, since it reflects whether the UAC is sending over a flow for  
which out bound processing is being applied, as opposed to merely  
indicating the UAC is capable of requesting outbound.

thanks,
-rohan

> thanks,
> -rohan
>
>
> On Jul 19, 2007, at 5:05 PM, Adam Roach wrote:
>> At the risk of further bloodying the corpse of this poor horse, I  
>> still think the usage of feature tags in outbound is way, way off  
>> in terms of the semantics described in 3261.
>>
>> Reviewing the way this is defined to work in SIP:
>>
>> 3261:8.1.1.9:
>>>    If the UAC supports extensions to SIP that can be applied by the
>>>    server to the response, the UAC SHOULD include a Supported header
>>>    field in the request listing the option tags (Section 19.2)  
>>> for those
>>>    extensions.
>>
>> 3261:8.2.4:
>>
>>>    A UAS that wishes to apply some extension when generating the
>>>    response MUST NOT do so unless support for that extension is
>>>    indicated in the Supported header field in the request.  If the
>>>    desired extension is not supported, the server SHOULD rely  
>>> only on
>>>    baseline SIP and any other extensions supported by the client.
>>> ...
>>>    Any extensions applied to a non-421 response MUST be listed in a
>>>    Require header field included in the response.  Of course, the  
>>> server
>>>    MUST NOT apply extensions not listed in the Supported header  
>>> field in
>>>    the request.  As a result of this, the Require header field in a
>>>    response will only ever contain option tags defined in standards-
>>>    track RFCs.
>>
>> There are five normative statements in the text I quote above. The  
>> mechanism described in the outbound draft actually manages to  
>> violate _all_ _five_ of them.
>>
>> The registrar infers *both* support for outbound *and* a request  
>> to apply the outbound extension by the presence of the "reg-id"  
>> Contact header field parameter. In particular, the client is  
>> apparently NOT supposed to send a "Supported: outbound" header  
>> field. This appears to contravene the normative statement in  
>> 8.1.1.9 cited above.
>>
>> The registrar indicates that outbound procedures are being applied  
>> by including a "Supported" header field in the REGISTER response.  
>> (In 3261, "Supported" in a _response_ indicates that an extension  
>> _can_ _be_ supported in a future request, NOT that it is being  
>> applied to the current one -- sections 11 and 13.3.1.4 both talk  
>> about Supported usage in responses.)
>>
>> In other words, the registrar (acting as a UAS) is applying an  
>> extension when generating a response without that extension being  
>> indicated in the "Supported" header field in the request. Again,  
>> this directly contravenes two normative statements from section  
>> 8.2.4 (cited above).
>>
>> The registrar is further applying an extension to a non-421  
>> response that is not listed in a "Require" header field, in  
>> violation of the remaining normative statement in the portion of  
>> section 8.2.4 cited above.
>>
>> Why are we doing this so very, very wrong? It's almost as if we've  
>> gone out of our way to violate both the spirit AND the law of  
>> feature tag handling in as many ways as possible. Most annoyingly,  
>> we get exactly the desired behavior if we just do things the way  
>> they're described in 3261. To wit:
>>
>>    * If a client supports outbound, it includes "Supported: outbound"
>>      in all REGISTER requests, regardless of whether the specific
>>      request makes use of that extension.
>>
>>    * The client adds a "reg-id" header field parameter to any  
>> Contacts
>>      to which outbound processing is to apply.
>>
>>    * The registrar includes "Require: outbound" in any REGISTER
>>      requests to which outbound behavior is being applied.
>>
>>    * The registrar is free to include "Supported: outbound" in any
>>      responses it generates -- and it means exactly what 3261 intends
>>      it to mean: that outbound is *supported*, but is not being  
>> applied
>>      to the response in question.
>>
>> /a



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From sip-bounces@ietf.org Fri Jul 20 13:32:56 2007
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Subject: [Sip] RE: Question about keep-alives in Outbound draft
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Moving to proper mailing list (i.e., sip).

Rohan, I also think that the draft could be a little bit clearer
about where the CRLF is sent, and where it stops, but I think it's a
consequence of how section 3 is structured, and the terms we use.

If we made the concept of "Outbound proxy" a bit clearer in the
document, and
use it throughout, and specifically say that the CRLF is sent to the
outbound proxy,=20
it would read much better.=20

Section 3 and above don't even use the term "outbound proxy" and instead
uses
euphemisms such as "edge proxy", "proxy", "registrar", "host" and
"server".

Furthermore, it starts with a special case (co-located Registrar and
Proxy),
and finishes with the the general case (outbound proxy is separate
from the Registrar).

I would suggest the following editorial changes:

- Move section 3.4 to where 3.2 currently is.

  - Remove the word "Proxy" in the box that says "Registrar/Proxy"
(i.e.,=20
    it should just say "Registrar". There is no need for having a proxy
    necessarily co-located in the Registrar, and it's immaterial to this
    draft.

   - The box that say "Edge1" and "Edge2" should really be labelled=20
     "Outbound Proxy  1" and "Oubound Proxy 2". The text should also
     use the term "outbound proxy" instead of "edge proxy", and it
     should just explain that often this configuration is called=20
     and "edge proxy".

- In section 3.2 (which would now be 3.3), clarify that this is just a=20
  special case of the outbound proxy and Registrar being co-located.
  Also use the term "outbound proxy" in the diagram and the text, as
  appropriate.

- In section 3.3 (which would now be 3.4), clarify that the Host1 and
  Host2 are really the outbound proxy 1 and 2, in the diagram.

- In section 3.5, use the term "outbound proxy" instead of server
  when talking about the entity receiving the ping and sending the pong.
 =20
- Section 5 should use the term "Outbound proxy" instead of "Edge
proxy".

- Section 5.4 is where we should say the "oubound proxy" does not=20
  forward the keep alives.

- Section 9 should use the terms "Primary outbound proxy" and "Secondary

  outbound proxy".

Cheers

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Friday, July 20, 2007 09:20
> To: Stucker, Brian (RICH1:AR00)
> Cc: Rohan Mahy; Cullen Jennings; sipping@ietf.org; Audet,=20
> Francois (SC100:3055)
> Subject: Re: Question about keep-alives in Outbound draft
>=20
>=20
> On Jul 19, 2007, at 9:50 PM, Brian Stucker wrote:
>=20
> > Cullen/Rohan,
> >
> > I have a question about keep-alive behavior in the -10=20
> version of the=20
> > outbound draft.
> >
> > How far are keep-alives propagated along the flow created at=20
> > registration?
> >
> > The text seems a bit hazy on how far it is expected for a=20
> keep- alive=20
> > to be propagated along the flow. Section 8 has text in the last=20
> > paragraph on page 25 that seems to imply that STUN=20
> keep-alives go no=20
> > further than the edge proxy immediately adjacent to the=20
> client device.=20
> > Section
> > 4.4.3
> > seems to also limit generation of STUN keep-alives to the=20
> User-Agent=20
> > only.
>=20
> one-hop.  one-hop only Mister Kamarov.
>=20
> > Likewise, section 4.4.1 and 5.4 seem to limit the TCP CRLF=20
> response to=20
> > the first edge-proxy as well.
>=20
> yes.
>=20
> I don't think either of these statements is hazy.  Can you=20
> propose a rewording?
>=20
> > Inclusion of the keep-stun or keep-crlf parameters for a Path URI=20
> > inserted by an edge proxy or registrar outside of the one in the=20
> > outbound-proxy-set seems moot as earlier edge-proxies that don't=20
> > support these mechanisms would ostensibly drop the keep-alives.
>=20
> if these parameters find their way into a URI handled by=20
> another proxy (instead of the UAC), they will be ignored (and=20
> no keepalives will be generated by the proxy).
>=20
> > Further, you only show a NAT/FW between the first=20
> edge-proxy and the=20
> > User-Agent in the figure for section 3.4.
>=20
> yes.  that was part of our design assumptions.  we had a hum=20
> on that about a year ago.  We assume that there is a full SIP=20
> connectivity among any of the proxies and that anything=20
> necessary to make that work is taken care of.
>=20
> > However, the spec never really
> > says very clearly what should occur at the edge-proxy:=20
> should it check=20
> > for freshness of the next-hop towards the registrar or not,=20
> based upon=20
> > what it observed in the Path header coming back in the registration=20
> > response?
>=20
> It says very clearly what to actually do and I hope it=20
> doesn't even hint that you should even begin to think about=20
> such a heinous thing as sending keepalives to the registrar=20
> from the EP.
>=20
> > Is there a reason that edge-proxies are not expected to regenerate/=20
> > relay the appropriate keep-alive message to the next entry=20
> on the Path=20
> > vector?
> > Not doing so means that some other mechanism must be=20
> employed to keep=20
> > bindings/pinholes open between the various edge-proxies and the=20
> > registrar proxy.
>=20
> If that problem exists in your environment, it is completely=20
> out of the scope of outbound.  The spec is about routing=20
> traffic over outbound-initiated flows from the UA.
>=20
> > For instance, in IMS you can have a P-CSCF in the visited network=20
> > talking to an IBCF in the visited network that is connected=20
> to an IBCF=20
> > in the home network that finally terminates at the S-CSCF=20
> in the home=20
> > network of the user that is registering. A NAT/FW is likely=20
> to exist=20
> > between the UA and the visited P-CSCF as well as between=20
> the IBCF in=20
> > the visited network and the IBCF in the home network.
>=20
> How do SIP messages make it today from a P-CSCF to an S-SCSF?
>=20
> > Another possible example (I believe) would be in the case of p2psip=20
> > where you may have multiple hops between the client and the node=20
> > acting as a registrar for the DHT algorithm. Wouldn't you=20
> want to keep=20
> > the entire path to the node storing your contact information in the=20
> > DHT alive or is there some other property of the DHT algorithm that=20
> > ensures that this is always the case even though outbound=20
> only keeps=20
> > the last hop fresh?
>=20
> p2psip has some unique requirements.  I don't think outbound=20
> prevents p2psip from addressing their requirements, but they=20
> are certainly not the concerns that outbound intended to address.
>=20
> thanks,
> -rohan
>=20
>=20
> > Regards,
> > Brian
>=20
>=20


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From sip-bounces@ietf.org Fri Jul 20 13:57:15 2007
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From: Rohan Mahy <rohan@ekabal.com>
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Subject: [Sip] Re: Question about keep-alives in Outbound draft
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Fran=E7ois,

We started off using the term outbound proxy in previous versions of =20
the draft and many people complained that the term did not have a =20
meaningful or useful definition.  It was one of those "its what I =20
mean" definitions.  I think the definitions of edge proxy, =20
authoritative proxy, registrar, and UAC are pretty tight.  There =20
certainly might be some places where we say "host" or "client" or =20
"proxy" where we should be saying something more well defined, but if =20=

so, I would like to change them to the correct terms, not to outbound =20=

proxy".

AFAIK, going back to outbound proxy will be a big step backwards for =20
the document.  Also, the registrar and proxy together is the simplest =20=

case to explain and I think it is still the most common topology.

thanks,
-rohan


On Jul 20, 2007, at 10:32 AM, Francois Audet wrote:
> Moving to proper mailing list (i.e., sip).
>
> Rohan, I also think that the draft could be a little bit clearer
> about where the CRLF is sent, and where it stops, but I think it's a
> consequence of how section 3 is structured, and the terms we use.
>
> If we made the concept of "Outbound proxy" a bit clearer in the
> document, and
> use it throughout, and specifically say that the CRLF is sent to the
> outbound proxy,
> it would read much better.
>
> Section 3 and above don't even use the term "outbound proxy" and =20
> instead
> uses
> euphemisms such as "edge proxy", "proxy", "registrar", "host" and
> "server".
>
> Furthermore, it starts with a special case (co-located Registrar and
> Proxy),
> and finishes with the the general case (outbound proxy is separate
> from the Registrar).
>
> I would suggest the following editorial changes:
>
> - Move section 3.4 to where 3.2 currently is.
>
>   - Remove the word "Proxy" in the box that says "Registrar/Proxy"
> (i.e.,
>     it should just say "Registrar". There is no need for having a =20
> proxy
>     necessarily co-located in the Registrar, and it's immaterial to =20=

> this
>     draft.
>
>    - The box that say "Edge1" and "Edge2" should really be labelled
>      "Outbound Proxy  1" and "Oubound Proxy 2". The text should also
>      use the term "outbound proxy" instead of "edge proxy", and it
>      should just explain that often this configuration is called
>      and "edge proxy".
>
> - In section 3.2 (which would now be 3.3), clarify that this is just a
>   special case of the outbound proxy and Registrar being co-located.
>   Also use the term "outbound proxy" in the diagram and the text, as
>   appropriate.
>
> - In section 3.3 (which would now be 3.4), clarify that the Host1 and
>   Host2 are really the outbound proxy 1 and 2, in the diagram.
>
> - In section 3.5, use the term "outbound proxy" instead of server
>   when talking about the entity receiving the ping and sending the =20
> pong.
>
> - Section 5 should use the term "Outbound proxy" instead of "Edge
> proxy".
>
> - Section 5.4 is where we should say the "oubound proxy" does not
>   forward the keep alives.
>
> - Section 9 should use the terms "Primary outbound proxy" and =20
> "Secondary
>
>   outbound proxy".
>
> Cheers
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: Friday, July 20, 2007 09:20
>> To: Stucker, Brian (RICH1:AR00)
>> Cc: Rohan Mahy; Cullen Jennings; sipping@ietf.org; Audet,
>> Francois (SC100:3055)
>> Subject: Re: Question about keep-alives in Outbound draft
>>
>>
>> On Jul 19, 2007, at 9:50 PM, Brian Stucker wrote:
>>
>>> Cullen/Rohan,
>>>
>>> I have a question about keep-alive behavior in the -10
>> version of the
>>> outbound draft.
>>>
>>> How far are keep-alives propagated along the flow created at
>>> registration?
>>>
>>> The text seems a bit hazy on how far it is expected for a
>> keep- alive
>>> to be propagated along the flow. Section 8 has text in the last
>>> paragraph on page 25 that seems to imply that STUN
>> keep-alives go no
>>> further than the edge proxy immediately adjacent to the
>> client device.
>>> Section
>>> 4.4.3
>>> seems to also limit generation of STUN keep-alives to the
>> User-Agent
>>> only.
>>
>> one-hop.  one-hop only Mister Kamarov.
>>
>>> Likewise, section 4.4.1 and 5.4 seem to limit the TCP CRLF
>> response to
>>> the first edge-proxy as well.
>>
>> yes.
>>
>> I don't think either of these statements is hazy.  Can you
>> propose a rewording?
>>
>>> Inclusion of the keep-stun or keep-crlf parameters for a Path URI
>>> inserted by an edge proxy or registrar outside of the one in the
>>> outbound-proxy-set seems moot as earlier edge-proxies that don't
>>> support these mechanisms would ostensibly drop the keep-alives.
>>
>> if these parameters find their way into a URI handled by
>> another proxy (instead of the UAC), they will be ignored (and
>> no keepalives will be generated by the proxy).
>>
>>> Further, you only show a NAT/FW between the first
>> edge-proxy and the
>>> User-Agent in the figure for section 3.4.
>>
>> yes.  that was part of our design assumptions.  we had a hum
>> on that about a year ago.  We assume that there is a full SIP
>> connectivity among any of the proxies and that anything
>> necessary to make that work is taken care of.
>>
>>> However, the spec never really
>>> says very clearly what should occur at the edge-proxy:
>> should it check
>>> for freshness of the next-hop towards the registrar or not,
>> based upon
>>> what it observed in the Path header coming back in the registration
>>> response?
>>
>> It says very clearly what to actually do and I hope it
>> doesn't even hint that you should even begin to think about
>> such a heinous thing as sending keepalives to the registrar
>> from the EP.
>>
>>> Is there a reason that edge-proxies are not expected to regenerate/
>>> relay the appropriate keep-alive message to the next entry
>> on the Path
>>> vector?
>>> Not doing so means that some other mechanism must be
>> employed to keep
>>> bindings/pinholes open between the various edge-proxies and the
>>> registrar proxy.
>>
>> If that problem exists in your environment, it is completely
>> out of the scope of outbound.  The spec is about routing
>> traffic over outbound-initiated flows from the UA.
>>
>>> For instance, in IMS you can have a P-CSCF in the visited network
>>> talking to an IBCF in the visited network that is connected
>> to an IBCF
>>> in the home network that finally terminates at the S-CSCF
>> in the home
>>> network of the user that is registering. A NAT/FW is likely
>> to exist
>>> between the UA and the visited P-CSCF as well as between
>> the IBCF in
>>> the visited network and the IBCF in the home network.
>>
>> How do SIP messages make it today from a P-CSCF to an S-SCSF?
>>
>>> Another possible example (I believe) would be in the case of p2psip
>>> where you may have multiple hops between the client and the node
>>> acting as a registrar for the DHT algorithm. Wouldn't you
>> want to keep
>>> the entire path to the node storing your contact information in the
>>> DHT alive or is there some other property of the DHT algorithm that
>>> ensures that this is always the case even though outbound
>> only keeps
>>> the last hop fresh?
>>
>> p2psip has some unique requirements.  I don't think outbound
>> prevents p2psip from addressing their requirements, but they
>> are certainly not the concerns that outbound intended to address.
>>
>> thanks,
>> -rohan
>>
>>
>>> Regards,
>>> Brian
>>
>>



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Subject: [Sip] Progressing ICE
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[This note is sent to MMUSIC, with bcc to BEHAVE, SIP, and P2PSIP =20
since the discussion has included those groups. Replies to MMUSIC =20
only please, as ICE is an MMUSIC working group draft]


The MMUSIC chairs have reviewed the mailing list discussion =20
pertaining to ICE over the last few days. While we agree that further =20=

implementation experience and publication of deployment results would =20=

be valuable, as it is for any new protocol, we do not believe this =20
sufficient cause to revisit the consensus to request publication of =20
draft-ietf-mmusic-ice-17.txt as a Proposed Standard.

The characteristics of Proposed Standard specifications are described =20=

in RFC 2026 section 4.1.1 as follows:

>    The entry-level maturity for the standards track is "Proposed
>    Standard".  A specific action by the IESG is required to move a
>    specification onto the standards track at the "Proposed Standard"
>    level.
>
>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has =20
> received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.  However, such experience is highly desirable, and will
>    usually represent a strong argument in favor of a Proposed Standard
>    designation.
>
>    The IESG may require implementation and/or operational experience
>    prior to granting Proposed Standard status to a specification that
>    materially affects the core Internet protocols or that specifies
>    behavior that may have significant operational impact on the
>    Internet.
>
>    A Proposed Standard should have no known technical omissions with
>    respect to the requirements placed upon it.  However, the IESG may
>    waive this requirement in order to allow a specification to advance
>    to the Proposed Standard state when it is considered to be =20
> useful and
>    necessary (and timely) even with known technical omissions.
>
>    Implementors should treat Proposed Standards as immature
>    specifications.  It is desirable to implement them in order to gain
>    experience and to validate, test, and clarify the specification.
>    However, since the content of Proposed Standards may be changed if
>    problems are found or better solutions are identified, deploying
>    implementations of such standards into a disruption-sensitive
>    environment is not recommended.

Our judgement is that ICE meets these requirements, and that there is =20=

general working group consensus to request publication. Accordingly, =20
our intention is to proceed as we previously announced, and request =20
that ICE be published as a Proposed Standard RFC.

Colin Perkins, J=F6rg Ott, and Jean-Fran=E7ois Mul=E9
MMUSIC chairs






_______________________________________________
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From sip-bounces@ietf.org Fri Jul 20 14:09:38 2007
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From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
To: David Barrett <dbarrett@quinthar.com>, 'Melinda Shore' <mshore@cisco.com>, 
	'Philip Matthews' <philip_matthews@magma.ca>
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David,

90% success in connection rate is very impressive. Are you refering to UDP, TCP
or both? As you say, fallback such as relay is essential for 100% connectivity.
One cannot use hole punching without a good fallback scheme.
 
regards,
suresh

--- David Barrett <dbarrett@quinthar.com> wrote:

> > -----Original Message-----
> > From: Melinda Shore [mailto:mshore@cisco.com]
> > 
> > ICE makes me want to pound red-hot nails into my eye sockets.  That said,
> > 90% really isn't nearly good enough for residential deployments.
> 
> Incidentally, 90% is where my stats come out today -- this doesn't include
> restricted-cone/symmetric and symmetric/symmetric penetration, which has
> been prototyped but not added to the production system yet.  The hope is
> this will get us much close to the 99% range, but it remains to be seen by
> how much.
> 
> Furthermore, I'd wager that for most applications (such as VoIP), even
> routing 1 in 10 calls through a central relay isn't going to break the bank.
> (Or for mega-discounts, use the full-cone and non-NAT'd clients (>30%) to
> act as "free" relays for their peers.)
> 
> Thus while NAT penetration using what I guess we're calling the "hole
> punching technique" has been demonstrated to work at 90%+ on a very large
> scale (hundreds of thousands of clients tested) we should expect that it's
> possible to get even better still.  If ICE can show how it'd do better,
> that'd be great.  But we can't tolerate ICE being any worse than that, and
> thus before rushing ahead with standardizing an unproven ICE, we should just
> take the time to prove it.
> 
> -david
> 
> 





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



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[This note is sent to MMUSIC, with bcc to BEHAVE, SIP, and P2PSIP =20
since the discussion has included those groups. Replies to MMUSIC =20
only please, as ICE is an MMUSIC working group draft]


The MMUSIC chairs have reviewed the mailing list discussion =20
pertaining to ICE over the last few days. While we agree that further =20=

implementation experience and publication of deployment results would =20=

be valuable, as it is for any new protocol, we do not believe this =20
sufficient cause to revisit the consensus to request publication of =20
draft-ietf-mmusic-ice-17.txt as a Proposed Standard.

The characteristics of Proposed Standard specifications are described =20=

in RFC 2026 section 4.1.1 as follows:

>    The entry-level maturity for the standards track is "Proposed
>    Standard".  A specific action by the IESG is required to move a
>    specification onto the standards track at the "Proposed Standard"
>    level.
>
>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has =20
> received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.  However, such experience is highly desirable, and will
>    usually represent a strong argument in favor of a Proposed Standard
>    designation.
>
>    The IESG may require implementation and/or operational experience
>    prior to granting Proposed Standard status to a specification that
>    materially affects the core Internet protocols or that specifies
>    behavior that may have significant operational impact on the
>    Internet.
>
>    A Proposed Standard should have no known technical omissions with
>    respect to the requirements placed upon it.  However, the IESG may
>    waive this requirement in order to allow a specification to advance
>    to the Proposed Standard state when it is considered to be =20
> useful and
>    necessary (and timely) even with known technical omissions.
>
>    Implementors should treat Proposed Standards as immature
>    specifications.  It is desirable to implement them in order to gain
>    experience and to validate, test, and clarify the specification.
>    However, since the content of Proposed Standards may be changed if
>    problems are found or better solutions are identified, deploying
>    implementations of such standards into a disruption-sensitive
>    environment is not recommended.

Our judgement is that ICE meets these requirements, and that there is =20=

general working group consensus to request publication. Accordingly, =20
our intention is to proceed as we previously announced, and request =20
that ICE be published as a Proposed Standard RFC.

Colin Perkins, J=F6rg Ott, and Jean-Fran=E7ois Mul=E9
MMUSIC chairs






_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Fri Jul 20 14:09:38 2007
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From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: RE: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
To: David Barrett <dbarrett@quinthar.com>, 'Melinda Shore' <mshore@cisco.com>, 
	'Philip Matthews' <philip_matthews@magma.ca>
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David,

90% success in connection rate is very impressive. Are you refering to UDP, TCP
or both? As you say, fallback such as relay is essential for 100% connectivity.
One cannot use hole punching without a good fallback scheme.
 
regards,
suresh

--- David Barrett <dbarrett@quinthar.com> wrote:

> > -----Original Message-----
> > From: Melinda Shore [mailto:mshore@cisco.com]
> > 
> > ICE makes me want to pound red-hot nails into my eye sockets.  That said,
> > 90% really isn't nearly good enough for residential deployments.
> 
> Incidentally, 90% is where my stats come out today -- this doesn't include
> restricted-cone/symmetric and symmetric/symmetric penetration, which has
> been prototyped but not added to the production system yet.  The hope is
> this will get us much close to the 99% range, but it remains to be seen by
> how much.
> 
> Furthermore, I'd wager that for most applications (such as VoIP), even
> routing 1 in 10 calls through a central relay isn't going to break the bank.
> (Or for mega-discounts, use the full-cone and non-NAT'd clients (>30%) to
> act as "free" relays for their peers.)
> 
> Thus while NAT penetration using what I guess we're calling the "hole
> punching technique" has been demonstrated to work at 90%+ on a very large
> scale (hundreds of thousands of clients tested) we should expect that it's
> possible to get even better still.  If ICE can show how it'd do better,
> that'd be great.  But we can't tolerate ICE being any worse than that, and
> thus before rushing ahead with standardizing an unproven ICE, we should just
> take the time to prove it.
> 
> -david
> 
> 





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



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--- Philip Matthews <philip_matthews@magma.ca> wrote:

> 
> On 18-Jul-07, at 15:31 , Henry Sinnreich wrote:
> 
> > Phil,
> >
> >> In my view, ICE _is_ a "Hole Punching Technique".
> >
> > Yes, you absolutely right, though the devil is in the details.
> >
> > We cannot go here into a shoot-out between hole punching or other
> > mentioned on the list vs. ICE, but there seems to be a huge difference
> > in effectiveness and performance.
> 
> What precise algorithm are you talking about when you say this?
> As far as I am concerned, ICE is simply a well-documented hole  
> punching algorithm.
> It differs from others that I have seen written up in the following  
> ways:
> - ICE has the concept of peer reflexive addresses, which other hole- 
> punching techniques that I have seen do not;

[suresh] I dont know what you are talking about. AFAIK, Peer reflexive address
is same as the public endpoint assigned to a node behind NAT. How can that be
not part of HOLE-PUNCHING technique? 

> - ICE has the concept of controlling/controlled endpoint, which gives  
> the controller additional flexibility in choosing a path;

[suresh] HOLE-PUNCHING technique is based on communication along a control path
with external rendezvous server(s), which act as the key controller. The 
peering endpoints either make requests to punch holes or receiving requests to
help in hole-punching by a peer.  

> - ICE does NOT have the concept of port prediction, which some other  
> techniques do (e.g., Saikat's TCP work).

[suresh] Hole punchign technique is not complete by itself to setup
connections. It needs a fallback scheme. Port prediction and relays can be
effective fallback to hole punching.


regards,
suresh
> 
> Philip
> 





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Cullen Jennings wrote:

> I would love to see more open source ICE code (there is lots already)  


Could you point me to the _lots_ of open source ICE code? Is there any 
open source implementation which supports TURN?


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Cullen Jennings wrote:

> I would love to see more open source ICE code (there is lots already)  


Could you point me to the _lots_ of open source ICE code? Is there any 
open source implementation which supports TURN?


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On 7/19/07 10:19 AM, Frank W. Miller wrote:
> FM: While I applaud the time and effort that the large vendors have
> obviously invested in this, this is a qualitative statement.  Perhaps Cisco
> and/or Microsoft have empirical data they would be willing to share about
> robustness and performance?

That's the kind of data you get between "proposed standard" and "draft 
standard" -- not *prior* to publication. I'm not sure why there's so 
much noise about setting the bar much, much higher for ICE than we do 
for other IETF proposals.

To be clear: to get this kind of experience (especially as it pertains 
to interoperability), you need to have a stable (i.e., frozen) 
specification. Internet Drafts simply don't provide an appropriate basis 
for the kinds of proof that you and others are demanding.

I think we would all benefit from a quick review of RFC 2026:

>    A Proposed Standard specification is generally stable, has resolved
>    known design choices, is believed to be well-understood, has received
>    significant community review, and appears to enjoy enough community
>    interest to be considered valuable.  However, further experience
>    might result in a change or even retraction of the specification
>    before it advances.
>
>    Usually, neither implementation nor operational experience is
>    required for the designation of a specification as a Proposed
>    Standard.
...
>    Implementors should treat Proposed Standards as immature
>    specifications.  It is desirable to implement them in order to gain
>    experience and to validate, test, and clarify the specification.
>    However, since the content of Proposed Standards may be changed if
>    problems are found or better solutions are identified, deploying
>    implementations of such standards into a disruption-sensitive
>    environment is not recommended.


/a


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The "Registrar" and "Authoritative proxy" definitions are prety tight, =
sure. That's
OK with me.

The problem is that you have no consistent term in the document for "the =
place where
the UAC sends both requests and keep alives".

In some places it's called "host", in others "server", in others "edge =
proxy",
in others "registrar", depending on topology. In other places, it's just =

avoided altogether, e.g.., the request is "sent" as oppose to the =
request is
"sent to edge proxy".

If we don't want to use the term "outbound proxy" that is used =
consistently in
RFC 3261 (presumably because it's both an "outbound" and an "inbound" =
proxy at the same=20
time, and because it means "the outbound proxy that conforms to the this =
draft"), then let's=20
use "Edge proxy" consistently. And we should state that "An Edge proxy =
is an Outbound
Proxy (as per RFC 3261) that conforms to the requirements of this =
specification" (if the
statement is true).

That would mean that in your co-located scenario, it would be a =
"Registrar/Edge Proxy"
instead of "Registrar/Proxy", and in your multiple connection scenario, =
it would be
"Edge Proxy 1" and "Edge Proxy 2".

Would that work?

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Friday, July 20, 2007 10:57
> To: Audet, Francois (SC100:3055)
> Cc: Rohan Mahy; Stucker, Brian (RICH1:AR00); Cullen Jennings;=20
> sip@ietf.org
> Subject: Re: Question about keep-alives in Outbound draft
>=20
> Fran=E7ois,
>=20
> We started off using the term outbound proxy in previous=20
> versions of the draft and many people complained that the=20
> term did not have a meaningful or useful definition.  It was=20
> one of those "its what I mean" definitions.  I think the=20
> definitions of edge proxy, authoritative proxy, registrar,=20
> and UAC are pretty tight.  There certainly might be some=20
> places where we say "host" or "client" or "proxy" where we=20
> should be saying something more well defined, but if so, I=20
> would like to change them to the correct terms, not to=20
> outbound proxy".
>=20
> AFAIK, going back to outbound proxy will be a big step=20
> backwards for the document.  Also, the registrar and proxy=20
> together is the simplest case to explain and I think it is=20
> still the most common topology.
>=20
> thanks,
> -rohan
>=20
>=20
> On Jul 20, 2007, at 10:32 AM, Francois Audet wrote:
> > Moving to proper mailing list (i.e., sip).
> >
> > Rohan, I also think that the draft could be a little bit=20
> clearer about=20
> > where the CRLF is sent, and where it stops, but I think it's a=20
> > consequence of how section 3 is structured, and the terms we use.
> >
> > If we made the concept of "Outbound proxy" a bit clearer in the=20
> > document, and use it throughout, and specifically say that=20
> the CRLF is=20
> > sent to the outbound proxy, it would read much better.
> >
> > Section 3 and above don't even use the term "outbound proxy" and=20
> > instead uses euphemisms such as "edge proxy", "proxy", "registrar",=20
> > "host" and "server".
> >
> > Furthermore, it starts with a special case (co-located=20
> Registrar and=20
> > Proxy), and finishes with the the general case (outbound proxy is=20
> > separate from the Registrar).
> >
> > I would suggest the following editorial changes:
> >
> > - Move section 3.4 to where 3.2 currently is.
> >
> >   - Remove the word "Proxy" in the box that says "Registrar/Proxy"
> > (i.e.,
> >     it should just say "Registrar". There is no need for having a=20
> > proxy
> >     necessarily co-located in the Registrar, and it's immaterial to=20
> > this
> >     draft.
> >
> >    - The box that say "Edge1" and "Edge2" should really be labelled
> >      "Outbound Proxy  1" and "Oubound Proxy 2". The text should also
> >      use the term "outbound proxy" instead of "edge proxy", and it
> >      should just explain that often this configuration is called
> >      and "edge proxy".
> >
> > - In section 3.2 (which would now be 3.3), clarify that=20
> this is just a
> >   special case of the outbound proxy and Registrar being co-located.
> >   Also use the term "outbound proxy" in the diagram and the text, as
> >   appropriate.
> >
> > - In section 3.3 (which would now be 3.4), clarify that the=20
> Host1 and
> >   Host2 are really the outbound proxy 1 and 2, in the diagram.
> >
> > - In section 3.5, use the term "outbound proxy" instead of server
> >   when talking about the entity receiving the ping and sending the=20
> > pong.
> >
> > - Section 5 should use the term "Outbound proxy" instead of "Edge=20
> > proxy".
> >
> > - Section 5.4 is where we should say the "oubound proxy" does not
> >   forward the keep alives.
> >
> > - Section 9 should use the terms "Primary outbound proxy" and=20
> > "Secondary
> >
> >   outbound proxy".
> >
> > Cheers
> >
> >> -----Original Message-----
> >> From: Rohan Mahy [mailto:rohan@ekabal.com]
> >> Sent: Friday, July 20, 2007 09:20
> >> To: Stucker, Brian (RICH1:AR00)
> >> Cc: Rohan Mahy; Cullen Jennings; sipping@ietf.org; Audet, Francois=20
> >> (SC100:3055)
> >> Subject: Re: Question about keep-alives in Outbound draft
> >>
> >>
> >> On Jul 19, 2007, at 9:50 PM, Brian Stucker wrote:
> >>
> >>> Cullen/Rohan,
> >>>
> >>> I have a question about keep-alive behavior in the -10
> >> version of the
> >>> outbound draft.
> >>>
> >>> How far are keep-alives propagated along the flow created at=20
> >>> registration?
> >>>
> >>> The text seems a bit hazy on how far it is expected for a
> >> keep- alive
> >>> to be propagated along the flow. Section 8 has text in the last=20
> >>> paragraph on page 25 that seems to imply that STUN
> >> keep-alives go no
> >>> further than the edge proxy immediately adjacent to the
> >> client device.
> >>> Section
> >>> 4.4.3
> >>> seems to also limit generation of STUN keep-alives to the
> >> User-Agent
> >>> only.
> >>
> >> one-hop.  one-hop only Mister Kamarov.
> >>
> >>> Likewise, section 4.4.1 and 5.4 seem to limit the TCP CRLF
> >> response to
> >>> the first edge-proxy as well.
> >>
> >> yes.
> >>
> >> I don't think either of these statements is hazy.  Can you=20
> propose a=20
> >> rewording?
> >>
> >>> Inclusion of the keep-stun or keep-crlf parameters for a Path URI=20
> >>> inserted by an edge proxy or registrar outside of the one in the=20
> >>> outbound-proxy-set seems moot as earlier edge-proxies that don't=20
> >>> support these mechanisms would ostensibly drop the keep-alives.
> >>
> >> if these parameters find their way into a URI handled by another=20
> >> proxy (instead of the UAC), they will be ignored (and no=20
> keepalives=20
> >> will be generated by the proxy).
> >>
> >>> Further, you only show a NAT/FW between the first
> >> edge-proxy and the
> >>> User-Agent in the figure for section 3.4.
> >>
> >> yes.  that was part of our design assumptions.  we had a=20
> hum on that=20
> >> about a year ago.  We assume that there is a full SIP connectivity=20
> >> among any of the proxies and that anything necessary to make that=20
> >> work is taken care of.
> >>
> >>> However, the spec never really
> >>> says very clearly what should occur at the edge-proxy:
> >> should it check
> >>> for freshness of the next-hop towards the registrar or not,
> >> based upon
> >>> what it observed in the Path header coming back in the=20
> registration=20
> >>> response?
> >>
> >> It says very clearly what to actually do and I hope it=20
> doesn't even=20
> >> hint that you should even begin to think about such a=20
> heinous thing=20
> >> as sending keepalives to the registrar from the EP.
> >>
> >>> Is there a reason that edge-proxies are not expected to=20
> regenerate/=20
> >>> relay the appropriate keep-alive message to the next entry
> >> on the Path
> >>> vector?
> >>> Not doing so means that some other mechanism must be
> >> employed to keep
> >>> bindings/pinholes open between the various edge-proxies and the=20
> >>> registrar proxy.
> >>
> >> If that problem exists in your environment, it is=20
> completely out of=20
> >> the scope of outbound.  The spec is about routing traffic over=20
> >> outbound-initiated flows from the UA.
> >>
> >>> For instance, in IMS you can have a P-CSCF in the visited network=20
> >>> talking to an IBCF in the visited network that is connected
> >> to an IBCF
> >>> in the home network that finally terminates at the S-CSCF
> >> in the home
> >>> network of the user that is registering. A NAT/FW is likely
> >> to exist
> >>> between the UA and the visited P-CSCF as well as between
> >> the IBCF in
> >>> the visited network and the IBCF in the home network.
> >>
> >> How do SIP messages make it today from a P-CSCF to an S-SCSF?
> >>
> >>> Another possible example (I believe) would be in the case=20
> of p2psip=20
> >>> where you may have multiple hops between the client and the node=20
> >>> acting as a registrar for the DHT algorithm. Wouldn't you
> >> want to keep
> >>> the entire path to the node storing your contact=20
> information in the=20
> >>> DHT alive or is there some other property of the DHT=20
> algorithm that=20
> >>> ensures that this is always the case even though outbound
> >> only keeps
> >>> the last hop fresh?
> >>
> >> p2psip has some unique requirements.  I don't think=20
> outbound prevents=20
> >> p2psip from addressing their requirements, but they are=20
> certainly not=20
> >> the concerns that outbound intended to address.
> >>
> >> thanks,
> >> -rohan
> >>
> >>
> >>> Regards,
> >>> Brian
> >>
> >>
>=20
>=20


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Subject: Re: [Sip] Re: Outbound: Feature Tag (Ab)Use
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Rohan,

The flag is needed, but you could make it a header parameter rather than a 
URI parameter (both for Contact and for Path).

There is no need for it to appear in the request URI of mid-dialog requests. 
The EP can use its own convention to encode the flag, due to things like 
GRUU it cannot rely on the 'ob' parameter being in the request URI anyway.

Regards,
Jeroen

Rohan Mahy wrote:
> On Jul 20, 2007, at 9:06 AM, Rohan Mahy wrote:
>
>> Hi Adam,
>>
>> What can I say but "You're right".  Unless there is some
>> fundamental objection, I will make this change pronto to my working
>> copy.
>>
>> I believe this also means I can remove the 'ob' parameter from the
>> UAC's Contact header.
>
> On further thought, I realized that the 'ob' parameter is still
> needed, since it reflects whether the UAC is sending over a flow for
> which out bound processing is being applied, as opposed to merely
> indicating the UAC is capable of requesting outbound.
>
> thanks,
> -rohan
>



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From: Rohan Mahy <rohan@ekabal.com>
Date: Fri, 20 Jul 2007 11:53:28 -0700
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Subject: [Sip] Re: Question about keep-alives in Outbound draft
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Hi Francois,

I think calling the composed registrar and proxy an Edge Proxy will =20
be confusing for a lot of people, but if you have a few places in =20
mind and want to send example text for one or two of these I think =20
its worth considering the change.

thanks,
-rohan


On Jul 20, 2007, at 11:22 AM, Francois Audet wrote:
> The "Registrar" and "Authoritative proxy" definitions are prety =20
> tight, sure. That's
> OK with me.
>
> The problem is that you have no consistent term in the document for =20=

> "the place where
> the UAC sends both requests and keep alives".
>
> In some places it's called "host", in others "server", in others =20
> "edge proxy",
> in others "registrar", depending on topology. In other places, it's =20=

> just
> avoided altogether, e.g.., the request is "sent" as oppose to the =20
> request is
> "sent to edge proxy".
>
> If we don't want to use the term "outbound proxy" that is used =20
> consistently in
> RFC 3261 (presumably because it's both an "outbound" and an =20
> "inbound" proxy at the same
> time, and because it means "the outbound proxy that conforms to the =20=

> this draft"), then let's
> use "Edge proxy" consistently. And we should state that "An Edge =20
> proxy is an Outbound
> Proxy (as per RFC 3261) that conforms to the requirements of this =20
> specification" (if the
> statement is true).
>
> That would mean that in your co-located scenario, it would be a =20
> "Registrar/Edge Proxy"
> instead of "Registrar/Proxy", and in your multiple connection =20
> scenario, it would be
> "Edge Proxy 1" and "Edge Proxy 2".
>
> Would that work?
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: Friday, July 20, 2007 10:57
>> To: Audet, Francois (SC100:3055)
>> Cc: Rohan Mahy; Stucker, Brian (RICH1:AR00); Cullen Jennings;
>> sip@ietf.org
>> Subject: Re: Question about keep-alives in Outbound draft
>>
>> Fran=E7ois,
>>
>> We started off using the term outbound proxy in previous
>> versions of the draft and many people complained that the
>> term did not have a meaningful or useful definition.  It was
>> one of those "its what I mean" definitions.  I think the
>> definitions of edge proxy, authoritative proxy, registrar,
>> and UAC are pretty tight.  There certainly might be some
>> places where we say "host" or "client" or "proxy" where we
>> should be saying something more well defined, but if so, I
>> would like to change them to the correct terms, not to
>> outbound proxy".
>>
>> AFAIK, going back to outbound proxy will be a big step
>> backwards for the document.  Also, the registrar and proxy
>> together is the simplest case to explain and I think it is
>> still the most common topology.
>>
>> thanks,
>> -rohan
>>
>>
>> On Jul 20, 2007, at 10:32 AM, Francois Audet wrote:
>>> Moving to proper mailing list (i.e., sip).
>>>
>>> Rohan, I also think that the draft could be a little bit
>> clearer about
>>> where the CRLF is sent, and where it stops, but I think it's a
>>> consequence of how section 3 is structured, and the terms we use.
>>>
>>> If we made the concept of "Outbound proxy" a bit clearer in the
>>> document, and use it throughout, and specifically say that
>> the CRLF is
>>> sent to the outbound proxy, it would read much better.
>>>
>>> Section 3 and above don't even use the term "outbound proxy" and
>>> instead uses euphemisms such as "edge proxy", "proxy", "registrar",
>>> "host" and "server".
>>>
>>> Furthermore, it starts with a special case (co-located
>> Registrar and
>>> Proxy), and finishes with the the general case (outbound proxy is
>>> separate from the Registrar).
>>>
>>> I would suggest the following editorial changes:
>>>
>>> - Move section 3.4 to where 3.2 currently is.
>>>
>>>   - Remove the word "Proxy" in the box that says "Registrar/Proxy"
>>> (i.e.,
>>>     it should just say "Registrar". There is no need for having a
>>> proxy
>>>     necessarily co-located in the Registrar, and it's immaterial to
>>> this
>>>     draft.
>>>
>>>    - The box that say "Edge1" and "Edge2" should really be labelled
>>>      "Outbound Proxy  1" and "Oubound Proxy 2". The text should also
>>>      use the term "outbound proxy" instead of "edge proxy", and it
>>>      should just explain that often this configuration is called
>>>      and "edge proxy".
>>>
>>> - In section 3.2 (which would now be 3.3), clarify that
>> this is just a
>>>   special case of the outbound proxy and Registrar being co-located.
>>>   Also use the term "outbound proxy" in the diagram and the text, as
>>>   appropriate.
>>>
>>> - In section 3.3 (which would now be 3.4), clarify that the
>> Host1 and
>>>   Host2 are really the outbound proxy 1 and 2, in the diagram.
>>>
>>> - In section 3.5, use the term "outbound proxy" instead of server
>>>   when talking about the entity receiving the ping and sending the
>>> pong.
>>>
>>> - Section 5 should use the term "Outbound proxy" instead of "Edge
>>> proxy".
>>>
>>> - Section 5.4 is where we should say the "oubound proxy" does not
>>>   forward the keep alives.
>>>
>>> - Section 9 should use the terms "Primary outbound proxy" and
>>> "Secondary
>>>
>>>   outbound proxy".
>>>
>>> Cheers
>>>
>>>> -----Original Message-----
>>>> From: Rohan Mahy [mailto:rohan@ekabal.com]
>>>> Sent: Friday, July 20, 2007 09:20
>>>> To: Stucker, Brian (RICH1:AR00)
>>>> Cc: Rohan Mahy; Cullen Jennings; sipping@ietf.org; Audet, Francois
>>>> (SC100:3055)
>>>> Subject: Re: Question about keep-alives in Outbound draft
>>>>
>>>>
>>>> On Jul 19, 2007, at 9:50 PM, Brian Stucker wrote:
>>>>
>>>>> Cullen/Rohan,
>>>>>
>>>>> I have a question about keep-alive behavior in the -10
>>>> version of the
>>>>> outbound draft.
>>>>>
>>>>> How far are keep-alives propagated along the flow created at
>>>>> registration?
>>>>>
>>>>> The text seems a bit hazy on how far it is expected for a
>>>> keep- alive
>>>>> to be propagated along the flow. Section 8 has text in the last
>>>>> paragraph on page 25 that seems to imply that STUN
>>>> keep-alives go no
>>>>> further than the edge proxy immediately adjacent to the
>>>> client device.
>>>>> Section
>>>>> 4.4.3
>>>>> seems to also limit generation of STUN keep-alives to the
>>>> User-Agent
>>>>> only.
>>>>
>>>> one-hop.  one-hop only Mister Kamarov.
>>>>
>>>>> Likewise, section 4.4.1 and 5.4 seem to limit the TCP CRLF
>>>> response to
>>>>> the first edge-proxy as well.
>>>>
>>>> yes.
>>>>
>>>> I don't think either of these statements is hazy.  Can you
>> propose a
>>>> rewording?
>>>>
>>>>> Inclusion of the keep-stun or keep-crlf parameters for a Path URI
>>>>> inserted by an edge proxy or registrar outside of the one in the
>>>>> outbound-proxy-set seems moot as earlier edge-proxies that don't
>>>>> support these mechanisms would ostensibly drop the keep-alives.
>>>>
>>>> if these parameters find their way into a URI handled by another
>>>> proxy (instead of the UAC), they will be ignored (and no
>> keepalives
>>>> will be generated by the proxy).
>>>>
>>>>> Further, you only show a NAT/FW between the first
>>>> edge-proxy and the
>>>>> User-Agent in the figure for section 3.4.
>>>>
>>>> yes.  that was part of our design assumptions.  we had a
>> hum on that
>>>> about a year ago.  We assume that there is a full SIP connectivity
>>>> among any of the proxies and that anything necessary to make that
>>>> work is taken care of.
>>>>
>>>>> However, the spec never really
>>>>> says very clearly what should occur at the edge-proxy:
>>>> should it check
>>>>> for freshness of the next-hop towards the registrar or not,
>>>> based upon
>>>>> what it observed in the Path header coming back in the
>> registration
>>>>> response?
>>>>
>>>> It says very clearly what to actually do and I hope it
>> doesn't even
>>>> hint that you should even begin to think about such a
>> heinous thing
>>>> as sending keepalives to the registrar from the EP.
>>>>
>>>>> Is there a reason that edge-proxies are not expected to
>> regenerate/
>>>>> relay the appropriate keep-alive message to the next entry
>>>> on the Path
>>>>> vector?
>>>>> Not doing so means that some other mechanism must be
>>>> employed to keep
>>>>> bindings/pinholes open between the various edge-proxies and the
>>>>> registrar proxy.
>>>>
>>>> If that problem exists in your environment, it is
>> completely out of
>>>> the scope of outbound.  The spec is about routing traffic over
>>>> outbound-initiated flows from the UA.
>>>>
>>>>> For instance, in IMS you can have a P-CSCF in the visited network
>>>>> talking to an IBCF in the visited network that is connected
>>>> to an IBCF
>>>>> in the home network that finally terminates at the S-CSCF
>>>> in the home
>>>>> network of the user that is registering. A NAT/FW is likely
>>>> to exist
>>>>> between the UA and the visited P-CSCF as well as between
>>>> the IBCF in
>>>>> the visited network and the IBCF in the home network.
>>>>
>>>> How do SIP messages make it today from a P-CSCF to an S-SCSF?
>>>>
>>>>> Another possible example (I believe) would be in the case
>> of p2psip
>>>>> where you may have multiple hops between the client and the node
>>>>> acting as a registrar for the DHT algorithm. Wouldn't you
>>>> want to keep
>>>>> the entire path to the node storing your contact
>> information in the
>>>>> DHT alive or is there some other property of the DHT
>> algorithm that
>>>>> ensures that this is always the case even though outbound
>>>> only keeps
>>>>> the last hop fresh?
>>>>
>>>> p2psip has some unique requirements.  I don't think
>> outbound prevents
>>>> p2psip from addressing their requirements, but they are
>> certainly not
>>>> the concerns that outbound intended to address.
>>>>
>>>> thanks,
>>>> -rohan
>>>>
>>>>
>>>>> Regards,
>>>>> Brian
>>>>
>>>>
>>
>>



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>
>    This specification recommends support for 'multipart/mixed' and
>    'multipart/alternative'.  At present, there are no SIP extensions
>    that use different 'multipart' subtypes such as parallel [RFC2046],
>    digest [RFC2046], or related [RFC2387].

Actually, RFC 4662 uses multipart/related. It does, however, use option 
tags -- so we're in good shape there.

/a


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12 second proof point of why content type doesn't work.

So here comes an INVITE.  I am an INFO-friendly UAS.  I negotiate by using
Accept.

This INVITE has the line:
Accept: application/kpml-response+xml


Wow!  This is great!  I know the UAC *wants* DTMF.  Why?  Because it accepts
a DTMF content type.  Excellent - I can now send them DTMF.

The user enters a digit, and I send an INFO with type
application/kpml-response+xml.

UAC says, "What the heck was that?"


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[Moved to SIP by chair request]

MUMBLEBRATZ for UA-to-UA communication will always work if you own the
endpoints.  Might as well use INFO for MUMBLEBRATZ.

If one wants interoperable communication or if you want to deploy equipment
from more than one vendor, or at least not having to specify exact behavior
(i.e., creating your own proprietary protocol), INFO doesn't work, as
already documented.

For that matter, we can save a ton of messages if we drop all the pesky
negotiation in SIP.  Why bother if we just "know" what the endpoints will
accept.  We just need to start sending RTP and things will just work :-)


On 7/18/07 10:10 AM, "Jesske, R" <R.Jesske@t-com.net> wrote:

> Dear all,
> I'm concerned about the discussion here and the aim of the draft to restr=
ict
> the use of INFO only to SIP-T and Q-SIG.
>=20
> If you now change restrict the RFC2976 you will tear down each implementa=
tion
> already done based on RFC2976. Like for DTMF that is broadly used within =
the
> world. So you are against backward compatibility?
>=20
> The draft is claiming that INFO has to be routed over each proxy where th=
e
> INVITE is routed to. I do not know what's the problem with that. A proxy
> forwards and the question is how much load will this cause. UA's can sent
> re-INVITES or REGISTER, RE-REGISTER a couple of times and nobody claims t=
hat.
> INFO is used for UA to UA communication so is this not proper enough that=
 at
> least the UA's will know what to do with the MIME?
> INFO is ideal to sent information that has to correlate with the Dialog u=
sing
> other mechanisms like an 2nd stream like http or a MESSAGE or a other con=
trol
> protocol can not fulfil this. Even it could be get lost when you want to
> traverse several SIP domains or if you interwork with the PSTN/ISDN.
>=20
> The Example of DTMF for INFO can only be done at mid dialog.
>=20
> Using KPML --> We have minimum 4 more Messages. No additional Load?
>=20
>=20
>=20
> Best Regards
>=20
> Roland
>=20
>=20
>=20
>> -----Urspr=FCngliche Nachricht-----
>> Von: Christer Holmberg (JO/LMF)
>> [mailto:christer.holmberg@ericsson.com]
>> Gesendet: Mittwoch, 18. Juli 2007 12:18
>> An: Dean Willis
>> Cc: IETF Sipping List
>> Betreff: RE: [Sipping] comments on draft-burger-sip-info-00
>>=20
>>=20
>>=20
>> Hi,=20
>>=20
>>>>>> - I disagree with the statement in chapter 1 regarding
>> the "context=20
>>>>>> for interpreting". The context is associated with the
>> content type.
>>>>>>=20
>>>>>=20
>>>>> Content type is, as I have repeatedly argued, completely
>> insufficient
>>=20
>>>>> for interpretation of context in the general case. What,
>> for example,
>>=20
>>>>> does it mean when an INFO shows up with a wav file
>> attached? Play it?
>>>>> Record it? Parse it using speech recognition and pass it
>> to a command
>>=20
>>>>> processor?
>>>>=20
>>>> The dialog (you can't send INFOs out-of-dialog, as far as I
>> know) will
>>=20
>>>> be associated with an application, and if the application
>> doesn't know
>>=20
>>>> what to do with it, it can promt the user: "You have now received a
>>>> file. What do you want to do with it? Save it? Play it? Display it?
>>>> Delete it?".
>>>=20
>>> And how does the receiver know what the sender intended?
>>>=20
>>> Assume you're taking to my "phone" application. You send digits.
>>> Should those be rendered to me, or fed into the user
>>> interface of a phone-based game we happen to be playing?
>>=20
>> How do you solve it with 2833?
>>=20
>> How do you solve it with KPML?
>>=20
>> You could even have the same problem with normal voice, e.g.
>> if the game
>> is controlled using voice recognition. Shall the received voice be fed
>> to the speaker, or fed to the game application?
>>=20
>> In some cases it may not matter if the data is fed to multiple
>> applications. But, if you do need to dispatch data so specific
>> applications you either need some kind of application
>> reference, or you
>> handle the dispatching locally in your device (the game
>> application can
>> tell it wants DTMFs, and/or the device chooses - either
>> automatically or
>> manually - which application will get the data etc). Or you use
>> different SIP sessions.
>>=20
>> In other cases you will not have the problem. For example,
>> videoControl
>> messages would normally be fed to a video control application, and
>> nowhere else.
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>>=20
>>=20
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>=20
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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ved this message in error, please immediately return this by email and then=
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[moved to SIP list at the request of the Chair]

In line.


On 7/16/07 7:43 AM, "Christer Holmberg (JO/LMF)"
<christer.holmberg@ericsson.com> wrote:

> 
> Hi,
> 
> I have a number off issues with the draft, and I think it's extremely
> un-objective.
> 
> - I disagree with the statement in chapter 1 regarding the "context for
> interpreting". The context is associated with the content type.
> 
> - I disagree with the statement in chapter 1 regarding not negotiation
> method. You can indicate what content types you support.
> 
> - I disagree with the stamement in chapter 1 saying that something would
> work only in limited or controlled networks. I am not really sure what a
> controlled network means in this context, though. Of course some people
> may have implemented something which only works between certain nodes,
> or in certain networks, but that's a separate issue (and not even
> specific to INFO).

See the trivial example where this breaks down that I just posted.  See also
RFC 3458 for a few examples of how media type alone is not sufficient for
determining why one is receiving the content.

> - I don't understand the statement in chapter 1 saying "Likewise, a
> routing proxy, such as the 3GPP IMS S-CSCF, will not expect to receive
> media server control messages.". A routing proxy most likely wouldn't
> care about media server control messages, so I don't understand the
> "will not expect" thing.

By expect, I meant from a traffic load perspective.  Clearly the S-CSCF will
be ignorant of most messages.
 
> - I don't understand the statement in chapter 2, saying "a DTMF digit
> type could be for user stimulus, post-dial digit collection, simple
> transport of a
> digit (no signaling context), or a mistake.". How is that specfic to
> INFO? I can press a DTMF button "by mistake" no matter what mechanism I
> use to transport it...

The mistake here is the INFO, not the digit press.  See the trivial example.
That is an example of a mistake.

> Now, these are issues Eric and I have agree to discuss off-line in
> Chicago.
> 
> I also have comments (some of which I've already given before) on
> Jonathan's statements:
> 
>> 1. confusion resulting from DTMF being sent in rfc2833 and INFO
> 
> I believe you have the same problem with KPML, and the solution is to
> use only one mechanism to send the DTMF.

And we agree :-)

>> 2. no way to signal whether the far end wants to receive DTMF via INFO;
> 
> The far end indicates what content types it supports for this session.
> 
>> it may be getting it via rfc2833 and that is fine for most cases
> 
> Fine. Then you don't indicate support for the content types used with
> the INFO solution

See trivial example of why this CANNOT work.

>> 3. proxies seeing INFO message load when they don't care about it
> 
> That is a potential issues, yes, but you MAY end up with exactly the
> same proxy path using KPML. It depends on the network architecture and
> routing mechanisms.

Yes, but at least KPML has a *few* throttling mechanisms, including digit
maps, the potential for intermediaries collapsing events, and the generic
RFC 3265 throttling mechanisms.  INFO has *NO* throttling mechanisms.

> And, with KPML you will end up with multiple dialogs. Whether that is a
> problem or not is for each one to decide, but I think it's fair to
> document that, and possible issues related to it.

Why we have multiple dialogs was beaten to death by RFC 3265.  What part of
unsolicited subscriptions is still OK?

>> 4. no way to pass digit maps so that you end up sending every digit in
> its own INFO or guess at the maps, so its inefficient
> 
> That is a limitation, yes, but it may not be required by everyone. And,
> it could be solved using a content type parameter.
> 
>> 5. security issues in case the DTMF is containing calling card info or
> credit cards - rfc2833 along with SRTP is much more secured
> 
> The security issue is not specific to DTMF, and using that argument we
> should send everything sensitive on the media plane. Also, in some
> environments the security may be provided other ways.
> 
>> and so on.
> 
> I don't have anything against documenting and discussing different
> issues. But we shall be fair, and look at issues with ALL alternatives.
> 
> The document doesn't say anything about WHY people are using INFO in the
> first place. It only says that "people think it's an IETF way of doing
> things", without considering that fact that there could be issues with
> other mechanisms why people are not using them.

Please enumerate why people use INFO, other than the RFC 4722, "It was the
only thing available" argument.


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[Please keep discussion of INFO to the SIP list.  Copied to the SIPPING list
because this message touches upon SPITSTOP.  Please keep discussion of
SPITSTOP to the SIPPING list.]


Let us take the case of Malicious Indicator.  This is where a subscriber
receives a call, realizes it is a malicious call (threatening, SPIT, etc.).
They then press the SPIT button (or press *xx), which tells their service
provider to mark the UAC as a bad actor.  One framework proposed for this is
the SPITSTOP Reference Scenario, draft-niccolini-sipping-spitstop-00.txt.

One might be tempted to think that INFO would be a great option for this
service.  It follows the return path of the INVITE, and so the INFO will hit
the caller's inbound proxy, which it can learn the caller is (statistically)
a bad actor.  That way the inbound proxy can do stuff like notify law
enforcement, add a vote to "this is a SPIT source," or other useful action.

However, consider a few issues.  First, since INFO lives exclusively within
an established dialog, there is no way to assert this message after the call
completes.  Second, this mechanism *relies* on an active service provider
topology.  If there is no proxy in the chain that will eat the INFO, the
caller will see the "this is a bad guy" message, which may have consequences
in the real world.  Third, there is no a'priori way for the UAS to know
whether or not it can issue the INFO.  The caller CERTAINLY will not
advertise, "please tell me if I am bad, particularly I know in advance that
I *am* a bad actor."

What is the correct way of doing this?  Here is where we have theory and
practice.

Theory says the proxy needs to SUBSCRIBE for the SPIT event at the UAS.  At
this point, life is good, interoperable, and works across networks.  This
enables events after the dialog is torn down, as presumably the SPIT event
will refer not to, "this dialog," which does not exist, but to "that dialog
identifier," which exists (and is theoretically unique) forever.

[PLEASE TURN YOUR FLAME THROWERS OFF AT THIS POINT]
Practice is that service providers might be able to add value by providing
proprietary phones or IAD's to their subscribers that just "know" they have
an implicit subscription for this service.  Yes, there is a whole host of
problems with this, but if you are in a controlled, limited, no desire for
inter-network connectivity, this mechanism will work.  Moreover, by
creating, in this case, a SPIT event package, it will even allow the
*possibility* of interoperable interworking if the endpoints implement the
full SUBSCRIBE protocol.

This approach shuts down the, "Oh, but with INFO I can save 3 messages, even
though this all happens after the call connects so it adds no user-visible
delay" argument.


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Just catching up :-(

One thing occurred to me - this document goes out of its way to say that 
a proxy need not do the loop check unless it forks. In general that 
makes sense, but in one case it does not:

Some proxies hide/suppress/obfuscate via headers, and then reverse the 
process on responses. A proxy that does so breaks loop checking by any 
of the proxies whose vias it hid. So I think a proxy that does this MUST 
perform a loop check itself even if it does not fork.

	Thanks,
	Paul


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=20
The last sentence of section 8 says "The SIP keepalive STUN usage
requires no backwards
compatibility.".

It's not clear "with what".

I'm assuming that you mean it works with rfc3489bis but not RFC 3489?

Please clarify (in the document and on the mailing list).


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(As WG co-chair)

You will not from the agenda that we have a number of author drafts to
be discussed.

In general we will be looking as part of this discussion to identify if
there is work to be performed within the WG on this subject.

The drafts are:

http://www.ietf.org/internet-drafts/draft-polk-sip-rph-in-responses-00.t
xt

Updates a SIP standards track document. Note that IEPREP are now out of
the requirements loop on this subject.

http://www.ietf.org/internet-drafts/draft-polk-sip-rph-new-namespaces-01
.txt

Prime function is the registration of new namespaces in the IANA
registry. The requirements for adding to this registry are: "A new
namespace MUST be defined in a Standards Track RFC, following the
'Standards Action' policy in [RFC2434]". The proposal is to take this
action using a SIP WG item.

http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01
.txt

An attempt to plug the underspecification of message body handling that
should have been in RFC 3261.

http://www.ietf.org/internet-drafts/draft-munakata-sip-privacy-new-01.tx
t

A replacement privacy mechanism for RFC 3323.

http://www.ietf.org/internet-drafts/draft-gurbani-sip-domain-certs-06.tx
t

Security document therefore requirements work is done in SIP.

http://www.ietf.org/internet-drafts/draft-dotson-sip-certificate-auth-03
.txt

Security document therefore requirements work is done in SIP.

http://www.ietf.org/internet-drafts/draft-burger-sip-info-00

An attempt to provider a sounder basis for INFO usage, and therefore an
reference point for questions that keep coming back to the SIP list.

http://www.ietf.org/internet-drafts/draft-wing-sip-identity-media-00.txt

Security document therefore requirements work is done in SIP.


Independently of what appears in the slides we will be looking for the
answers to two questions for each of these drafts:

Is there support for a body of work in this area?
-------------------------------------------------

To answer this question it is suggested that you primarily study the
abstract for the draft, and if necessary seek clarification on the list
or at the microphone in the meeting. Instead of this abstract, you may
wish to suggest some alternative scope for adoption. At this point we
don't necessarily need to decide whether the work is standards track,
BCP or informational although that will form a part of the discussions
the ADs and the chairs have in setting milestones. Note that answering
YES to this also expresses an intention within the WG to provide
resources to work on and review this draft (therefore do not answer
"YES" if you just consider it mostly harmless.

There are three answers to this question:

YES
NO
NEEDS FURTHER WORK BEFORE WE DECIDE

A yes result means that the chairs will discuss with the ADs whether it
is appropriate to set new milestones for this work and what they should
be, taking into account the discussion made. We will also proceed to the
second question.

Does the document currently provided constitute a baseline draft for the
identified work?
------------------------------------------------------------------------
-----------------

What the WG is looking for here is whether the document, or its next
revision, could (subject to the agreement of the milestones with the AD)
become a draft-ietf-sip-xxxx document. This doesn't mean it has to be
perfect, merely that this is a better starting point to the work than
some other draft that may exist or that someone intends to produce.

Again there are three answers to this question:

YES
NO
NEEDS FURTHER WORK BEFORE WE DECIDE

It is entirely appropriate to hold discussions on the list to formulate
your opinions to answer the above questions. Expressions of support are
also appropriate. Note that we may not proceed to the questions on all
these drafts if it becomes apparent from list discussion or discussion
at the mic that the answer will not be "YES".


Requests for clarification on any of the above are welcome.

Regards

Keith


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From sip-bounces@ietf.org Sat Jul 21 10:09:11 2007
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   Some proxies hide/suppress/obfuscate via headers, and then reverse the 
   process on responses. A proxy that does so breaks loop checking by any 
   of the proxies whose vias it hid. So I think a proxy that does this MUST 
   perform a loop check itself even if it does not fork.

Hmmm, if you have a loop consisting of two obfuscating non-forking
proxies and one non-obfuscating forking proxy, could any of the three
proxies detect the loop?

Dale


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From sip-bounces@ietf.org Sat Jul 21 10:48:07 2007
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From: Jiri Kuthan <jiri@iptel.org>
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At 20:05 19/07/2007, David Barrett wrote:
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment databefore
>> LC for RFC
>> 
>> So I think that the right course of action is that folks doubting ICE
>> reliability
>> make their case (by that I mean specific arguments rather than emotional
>> debates
>> about red-colored nails and other hardware).
>
>Er, my "specific argument" is "until it's been proven to work, it hasn't
>been proven to work".  It's tautological, but no less true for it.
>
>Indeed, if you're *not* doubting ICE, why aren't you?

Hi David,

I'm not entirely sure what is the step you are suggesting?

I think ICE is architecturally right answer to the right problem.
Notwithstanding that, there are certainly some things I prefer
differently (e.g., the infamous debate about proactive hole punching)
or may turn out to be bugs. I think though that the key problem at
this point is of institutional nature. In the IETF language, "hasn't
been proven to work" is adequate to "lacking running code" and this
appears to be a phenomen which is indeed occuring in the IETF and
is not unique to ICE.

The question is how to get this fixed?

Some other standardization bodies enjoying the amount of contributions
and complexity as the IETF is enjoying now (S in SIP does not really
stand for Simple), proceed with 'after-fact filters', in that they
produce the initial specs faster and later produce 'profiles' which
separate what has proved and what less. Plus -bis versions are produced
for whatever appears worth puttting effort in it (i.e., for which there
is running code and field evidence of improvement need).This could be 
an approach  which may be more feasible in today's IETF's dimension than 
trying to get back to historical roots.

-jiri 



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



From sip-bounces@ietf.org Sat Jul 21 11:00:19 2007
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(As SIP WG co-chair)

We have a few documents we are looking to get into WGLC in the near
future.

In parallel with the WGLC, we would like some dedicated reviewers who
will undertake to do a full review of the document. The purpose of this
mail is therefore to invite expressions of interest in performing a
review of either of the following documents:

http://www.ietf.org/internet-drafts/draft-ietf-sip-subnot-etags-00.txt

http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-03.
txt

(Note that for this latter document we will also be looking for some
cross WG review)


Further out, but one we know we will need a number of reviewers for
specific sections:

http://tools.ietf.org/id/draft-ietf-sip-sec-flows-01.txt

Please respond to the WG chairs.

Regards

Keith



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From sip-bounces@ietf.org Sat Jul 21 13:11:10 2007
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Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deploymentdatabefore LC
	for RFC
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From: "Henry Sinnreich" <hsinnrei@adobe.com>
To: "Jiri Kuthan" <jiri@iptel.org>, "David Barrett" <dbarrett@quinthar.com>,
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I fully agree with Jiri that this is an IETF problem and not an ICE
problem.
ICE-17 is a great document, no doubt about it.

What happens however of someone finds an innovative approach that may
work better? Will it be rejected because it will not be RFC standards
compliant?

What could an innovator tell customers about a solution that works
better but is not IETF ICE RFC standards compliant?

Henry

-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]=20
Sent: Saturday, July 21, 2007 9:48 AM
To: David Barrett; 'Jan Janak'; 'Juha Heinanen'
Cc: sip@ietf.org; mmusic@ietf.org
Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
deploymentdatabefore LC for RFC

At 20:05 19/07/2007, David Barrett wrote:
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment
databefore
>> LC for RFC
>>=20
>> So I think that the right course of action is that folks doubting ICE
>> reliability
>> make their case (by that I mean specific arguments rather than
emotional
>> debates
>> about red-colored nails and other hardware).
>
>Er, my "specific argument" is "until it's been proven to work, it
hasn't
>been proven to work".  It's tautological, but no less true for it.
>
>Indeed, if you're *not* doubting ICE, why aren't you?

Hi David,

I'm not entirely sure what is the step you are suggesting?

I think ICE is architecturally right answer to the right problem.
Notwithstanding that, there are certainly some things I prefer
differently (e.g., the infamous debate about proactive hole punching)
or may turn out to be bugs. I think though that the key problem at
this point is of institutional nature. In the IETF language, "hasn't
been proven to work" is adequate to "lacking running code" and this
appears to be a phenomen which is indeed occuring in the IETF and
is not unique to ICE.

The question is how to get this fixed?

Some other standardization bodies enjoying the amount of contributions
and complexity as the IETF is enjoying now (S in SIP does not really
stand for Simple), proceed with 'after-fact filters', in that they
produce the initial specs faster and later produce 'profiles' which
separate what has proved and what less. Plus -bis versions are produced
for whatever appears worth puttting effort in it (i.e., for which there
is running code and field evidence of improvement need).This could be=20
an approach  which may be more feasible in today's IETF's dimension than

trying to get back to historical roots.

-jiri=20



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


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



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To: Dale.Worley@comcast.net
Subject: Re: [Sip] Comment on draft-ietf-sip-fork-loop-fix-05
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Dale.Worley@comcast.net wrote:
>    From: Paul Kyzivat <pkyzivat@cisco.com>
> 
>    Some proxies hide/suppress/obfuscate via headers, and then reverse the 
>    process on responses. A proxy that does so breaks loop checking by any 
>    of the proxies whose vias it hid. So I think a proxy that does this MUST 
>    perform a loop check itself even if it does not fork.
> 
> Hmmm, if you have a loop consisting of two obfuscating non-forking
> proxies and one non-obfuscating forking proxy, could any of the three
> proxies detect the loop?

Yes, if the obfuscating proxy performs the loop checking based on its 
own Via.

But if you have *two* obfuscating proxies in the loop then I think it 
becomes impossible to detect the loop.

In a way, this is similar the a loop that includes both a sip leg and a 
PSTN leg.

I don't have a good solution for either of these.

	Paul


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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Hi,=20

>[Please keep discussion of INFO to the SIP list.  Copied to=20
>the SIPPING list because this message touches upon SPITSTOP. =20
>Please keep discussion of SPITSTOP to the SIPPING list.]
>=20
>=20
>Let us take the case of Malicious Indicator.  This is where a=20
>subscriber receives a call, realizes it is a malicious call=20
>(threatening, SPIT, etc.).
>They then press the SPIT button (or press *xx), which tells=20
>their service provider to mark the UAC as a bad actor.  One=20
>framework proposed for this is the SPITSTOP Reference=20
>Scenario, draft-niccolini-sipping-spitstop-00.txt.
>=20
>One might be tempted to think that INFO would be a great=20
>option for this service.  It follows the return path of the=20
>INVITE, and so the INFO will hit the caller's inbound proxy,=20
>which it can learn the caller is (statistically) a bad actor.=20
>That way the inbound proxy can do stuff like notify law=20
>enforcement, add a vote to "this is a SPIT source," or other=20
>useful action.
>=20
>However, consider a few issues.  First, since INFO lives=20
>exclusively within an established dialog, there is no way to=20
>assert this message after the call completes.  Second, this=20
>mechanism *relies* on an active service provider topology. =20
>If there is no proxy in the chain that will eat the INFO, the=20
>caller will see the "this is a bad guy" message, which may=20
>have consequences in the real world.  Third, there is no=20
>a'priori way for the UAS to know whether or not it can issue=20
>the INFO.  The caller CERTAINLY will not advertise, "please=20
>tell me if I am bad, particularly I know in advance that I=20
>*am* a bad actor."

This is an example of a special case, where you don't want to INFO to go
"all the way back", ie you want it to be terminated somewhere along the
dialog path, and I don't think that is a valid justification why you
couldn't use INFO in cases where you don't have that issue.

I don't even think this behavior would be allowed according to the proxy
rules, so the proxy would more or less be a B2BUA in this case. So, if
the outbound proxy supports this mechanism, and DOES want to use INFO,
it can insert application/mal-ind in the Accept header before forwarding
it towards the UAS (since it is acting as a B2BUA, it is allowed to do
so).
=20
>What is the correct way of doing this?  Here is where we have theory
and practice.
>=20
>Theory says the proxy needs to SUBSCRIBE for the SPIT event=20
>at the UAS.  At this point, life is good, interoperable, and=20
>works across networks.  This enables events after the dialog=20
>is torn down, as presumably the SPIT event will refer not to,=20
>"this dialog," which does not exist, but to "that dialog=20
>identifier," which exists (and is theoretically unique) forever.
>
>[PLEASE TURN YOUR FLAME THROWERS OFF AT THIS POINT] Practice=20
>is that service providers might be able to add value by=20
>providing proprietary phones or IAD's to their subscribers=20
>that just "know" they have an implicit subscription for this=20
>service.  Yes, there is a whole host of problems with this,=20
>but if you are in a controlled, limited, no desire for=20
>inter-network connectivity, this mechanism will work. =20
>Moreover, by creating, in this case, a SPIT event package, it=20
>will even allow the *possibility* of interoperable interworking if the
endpoints=20
>implement the full SUBSCRIBE protocol.
>=20
>This approach shuts down the, "Oh, but with INFO I can save 3 messages,
even though this all happens after the call=20
>connects so it adds no user-visible delay" argument.

I am a little confused. I thought YOU have been saying that we shall
define mechanisms which are NOT based on behavior in controlled and
limited networks...

Regards,

Christer


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Subject: RE: [Sip] INFO and Content Negotiation
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Hi,

I don't really understand your issue here... Did you send the text using =
INFO, or something? ;)

Regards,

Christer=20

> -----Original Message-----
> From: Eric Burger [mailto:eburger@bea.com]=20
> Sent: 20. hein=E4kuuta 2007 23:14
> To: IETF SIP
> Subject: [Sip] INFO and Content Negotiation
>=20
> 12 second proof point of why content type doesn't work.
>=20
> So here comes an INVITE.  I am an INFO-friendly UAS.  I=20
> negotiate by using Accept.
>=20
> This INVITE has the line:
> Accept: application/kpml-response+xml
>=20
>=20
> Wow!  This is great!  I know the UAC *wants* DTMF.  Why? =20
> Because it accepts a DTMF content type.  Excellent - I can=20
> now send them DTMF.
>=20
> The user enters a digit, and I send an INFO with type=20
> application/kpml-response+xml.
>=20
> UAC says, "What the heck was that?"
>=20
>=20
> Notice:  This email message, together with any attachments,=20
> may contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and  affiliated entities,  that may be=20
> confidential,  proprietary,  copyrighted  and/or legally=20
> privileged, and is intended solely for the use of the=20
> individual or entity named in this message. If you are not=20
> the intended recipient, and have received this message in=20
> error, please immediately return this by email and then delete it.
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] Re: AW: [Sipping] comments on draft-burger-sip-info-00
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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	"Jesske, R" <R.Jesske@t-com.net>
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Hi,
=20
>[Moved to SIP by chair request]
>=20
>MUMBLEBRATZ for UA-to-UA communication will always work if=20
>you own the endpoints.  Might as well use INFO for MUMBLEBRATZ.
>=20
>If one wants interoperable communication or if you want to=20
>deploy equipment from more than one vendor, or at least not=20
>having to specify exact behavior (i.e., creating your own=20
>proprietary protocol), INFO doesn't work, as already documented.

If you are going to use INFO to carry videoControl messages, of course =
those messages, and the meaning of those, must be defined and documented =
- in order to make them non-proprietary.

It's the same with ISUP. You use INFO to carry SPECIFIED ISUP messages, =
and endpoints supporting ISUP know the procedures how to deal with the =
messages, because it is defined in standard documentation.
=20
>For that matter, we can save a ton of messages if we drop all=20
>the pesky negotiation in SIP.  Why bother if we just "know"=20
>what the endpoints will accept.  We just need to start=20
>sending RTP and things will just work :-)

Please show me message you save by not using INFO. The negotiation can =
in most cases be "encapsulated" in SIP messages you are going to send =
anyway.

But, if you use an event based mechanism, you actually WILL get more SIP =
messages, because you need to send the NOTIFY request(s).

Regards,

Christer






>=20
>=20
> On 7/18/07 10:10 AM, "Jesske, R" <R.Jesske@t-com.net> wrote:
>=20
> > Dear all,
> > I'm concerned about the discussion here and the aim of the draft to=20
> > restrict the use of INFO only to SIP-T and Q-SIG.
> >=20
> > If you now change restrict the RFC2976 you will tear down each=20
> > implementation already done based on RFC2976. Like for DTMF that is=20
> > broadly used within the world. So you are against backward=20
> compatibility?
> >=20
> > The draft is claiming that INFO has to be routed over each=20
> proxy where=20
> > the INVITE is routed to. I do not know what's the problem=20
> with that. A=20
> > proxy forwards and the question is how much load will this=20
> cause. UA's=20
> > can sent re-INVITES or REGISTER, RE-REGISTER a couple of=20
> times and nobody claims that.
> > INFO is used for UA to UA communication so is this not=20
> proper enough=20
> > that at least the UA's will know what to do with the MIME?
> > INFO is ideal to sent information that has to correlate with the=20
> > Dialog using other mechanisms like an 2nd stream like http or a=20
> > MESSAGE or a other control protocol can not fulfil this.=20
> Even it could=20
> > be get lost when you want to traverse several SIP domains=20
> or if you interwork with the PSTN/ISDN.
> >=20
> > The Example of DTMF for INFO can only be done at mid dialog.
> >=20
> > Using KPML --> We have minimum 4 more Messages. No additional Load?
> >=20
> >=20
> >=20
> > Best Regards
> >=20
> > Roland
> >=20
> >=20
> >=20
> >> -----Urspr=FCngliche Nachricht-----
> >> Von: Christer Holmberg (JO/LMF)
> >> [mailto:christer.holmberg@ericsson.com]
> >> Gesendet: Mittwoch, 18. Juli 2007 12:18
> >> An: Dean Willis
> >> Cc: IETF Sipping List
> >> Betreff: RE: [Sipping] comments on draft-burger-sip-info-00
> >>=20
> >>=20
> >>=20
> >> Hi,
> >>=20
> >>>>>> - I disagree with the statement in chapter 1 regarding
> >> the "context
> >>>>>> for interpreting". The context is associated with the
> >> content type.
> >>>>>>=20
> >>>>>=20
> >>>>> Content type is, as I have repeatedly argued, completely
> >> insufficient
> >>=20
> >>>>> for interpretation of context in the general case. What,
> >> for example,
> >>=20
> >>>>> does it mean when an INFO shows up with a wav file
> >> attached? Play it?
> >>>>> Record it? Parse it using speech recognition and pass it
> >> to a command
> >>=20
> >>>>> processor?
> >>>>=20
> >>>> The dialog (you can't send INFOs out-of-dialog, as far as I
> >> know) will
> >>=20
> >>>> be associated with an application, and if the application
> >> doesn't know
> >>=20
> >>>> what to do with it, it can promt the user: "You have now=20
> received a=20
> >>>> file. What do you want to do with it? Save it? Play it?=20
> Display it?
> >>>> Delete it?".
> >>>=20
> >>> And how does the receiver know what the sender intended?
> >>>=20
> >>> Assume you're taking to my "phone" application. You send digits.
> >>> Should those be rendered to me, or fed into the user=20
> interface of a=20
> >>> phone-based game we happen to be playing?
> >>=20
> >> How do you solve it with 2833?
> >>=20
> >> How do you solve it with KPML?
> >>=20
> >> You could even have the same problem with normal voice, e.g.
> >> if the game
> >> is controlled using voice recognition. Shall the received voice be=20
> >> fed to the speaker, or fed to the game application?
> >>=20
> >> In some cases it may not matter if the data is fed to multiple=20
> >> applications. But, if you do need to dispatch data so specific=20
> >> applications you either need some kind of application=20
> reference, or=20
> >> you handle the dispatching locally in your device (the game=20
> >> application can tell it wants DTMFs, and/or the device chooses -=20
> >> either automatically or manually - which application will get the=20
> >> data etc). Or you use different SIP sessions.
> >>=20
> >> In other cases you will not have the problem. For example,=20
> >> videoControl messages would normally be fed to a video control=20
> >> application, and nowhere else.
> >>=20
> >> Regards,
> >>=20
> >> Christer
> >>=20
> >>=20
> >>=20
> >> _______________________________________________
> >> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >> This list is for NEW development of the application of SIP Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sip@ietf.org for new developments of core SIP
> >>=20
> >=20
> >=20
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sip@ietf.org for new developments of core SIP
>=20
>=20
> Notice:  This email message, together with any attachments,=20
> may contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and  affiliated entities,  that may be=20
> confidential,  proprietary,  copyrighted  and/or legally=20
> privileged, and is intended solely for the use of the=20
> individual or entity named in this message. If you are not=20
> the intended recipient, and have received this message in=20
> error, please immediately return this by email and then delete it.
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Sat Jul 21 16:00:12 2007
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Looks good to me.

I will make my pro-forma suggestion that the syntax make provision for 
parameters just in case there may be need for an extension in the future:

   Max-Breadth = "Max-Breadth" HCOLON 1*DIGIT *( SEMI generic-param )

	Paul


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From: "Jonathan Rosenberg \(jdrosen\)" <jdrosen@cisco.com>
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Of course they can bring such a thing to ietf. There is a long history =
of new work improving on the old (ice improving on stun, v6 improving on =
v4, ospf improving on rip, and so on).

I look forward to your draft suggesting something better.

Jonathan R.

 -----Original Message-----
From: 	Henry Sinnreich [mailto:hsinnrei@adobe.com]
Sent:	Saturday, July 21, 2007 01:11 PM Eastern Standard Time
To:	Jiri Kuthan; David Barrett; Jan Janak; Juha Heinanen
Cc:	sip@ietf.org; mmusic@ietf.org
Subject:	RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE =
deploymentdatabefore LCfor RFC

I fully agree with Jiri that this is an IETF problem and not an ICE
problem.
ICE-17 is a great document, no doubt about it.

What happens however of someone finds an innovative approach that may
work better? Will it be rejected because it will not be RFC standards
compliant?

What could an innovator tell customers about a solution that works
better but is not IETF ICE RFC standards compliant?

Henry

-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]=20
Sent: Saturday, July 21, 2007 9:48 AM
To: David Barrett; 'Jan Janak'; 'Juha Heinanen'
Cc: sip@ietf.org; mmusic@ietf.org
Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
deploymentdatabefore LC for RFC

At 20:05 19/07/2007, David Barrett wrote:
>> -----Original Message-----
>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment
databefore
>> LC for RFC
>>=20
>> So I think that the right course of action is that folks doubting ICE
>> reliability
>> make their case (by that I mean specific arguments rather than
emotional
>> debates
>> about red-colored nails and other hardware).
>
>Er, my "specific argument" is "until it's been proven to work, it
hasn't
>been proven to work".  It's tautological, but no less true for it.
>
>Indeed, if you're *not* doubting ICE, why aren't you?

Hi David,

I'm not entirely sure what is the step you are suggesting?

I think ICE is architecturally right answer to the right problem.
Notwithstanding that, there are certainly some things I prefer
differently (e.g., the infamous debate about proactive hole punching)
or may turn out to be bugs. I think though that the key problem at
this point is of institutional nature. In the IETF language, "hasn't
been proven to work" is adequate to "lacking running code" and this
appears to be a phenomen which is indeed occuring in the IETF and
is not unique to ICE.

The question is how to get this fixed?

Some other standardization bodies enjoying the amount of contributions
and complexity as the IETF is enjoying now (S in SIP does not really
stand for Simple), proceed with 'after-fact filters', in that they
produce the initial specs faster and later produce 'profiles' which
separate what has proved and what less. Plus -bis versions are produced
for whatever appears worth puttting effort in it (i.e., for which there
is running code and field evidence of improvement need).This could be=20
an approach  which may be more feasible in today's IETF's dimension than

trying to get back to historical roots.

-jiri=20



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


_______________________________________________
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<P><FONT SIZE=3D2>Of course they can bring such a thing to ietf. There =
is a long history of new work improving on the old (ice improving on =
stun, v6 improving on v4, ospf improving on rip, and so on).<BR>
<BR>
I look forward to your draft suggesting something better.<BR>
<BR>
Jonathan R.<BR>
<BR>
&nbsp;-----Original Message-----<BR>
From: &nbsp; Henry Sinnreich [<A =
HREF=3D"mailto:hsinnrei@adobe.com">mailto:hsinnrei@adobe.com</A>]<BR>
Sent:&nbsp;&nbsp; Saturday, July 21, 2007 01:11 PM Eastern Standard =
Time<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; Jiri Kuthan; David Barrett; Jan Janak; Juha =
Heinanen<BR>
Cc:&nbsp;&nbsp;&nbsp;&nbsp; sip@ietf.org; mmusic@ietf.org<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [MMUSIC] Re: =
[Sip] RE: [BEHAVE] Re: ICE deploymentdatabefore LCfor RFC<BR>
<BR>
I fully agree with Jiri that this is an IETF problem and not an ICE<BR>
problem.<BR>
ICE-17 is a great document, no doubt about it.<BR>
<BR>
What happens however of someone finds an innovative approach that =
may<BR>
work better? Will it be rejected because it will not be RFC =
standards<BR>
compliant?<BR>
<BR>
What could an innovator tell customers about a solution that works<BR>
better but is not IETF ICE RFC standards compliant?<BR>
<BR>
Henry<BR>
<BR>
-----Original Message-----<BR>
From: Jiri Kuthan [<A =
HREF=3D"mailto:jiri@iptel.org">mailto:jiri@iptel.org</A>]<BR>
Sent: Saturday, July 21, 2007 9:48 AM<BR>
To: David Barrett; 'Jan Janak'; 'Juha Heinanen'<BR>
Cc: sip@ietf.org; mmusic@ietf.org<BR>
Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE<BR>
deploymentdatabefore LC for RFC<BR>
<BR>
At 20:05 19/07/2007, David Barrett wrote:<BR>
&gt;&gt; -----Original Message-----<BR>
&gt;&gt; From: Jiri Kuthan [<A =
HREF=3D"mailto:jiri@iptel.org">mailto:jiri@iptel.org</A>]<BR>
&gt;&gt; Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE =
deployment<BR>
databefore<BR>
&gt;&gt; LC for RFC<BR>
&gt;&gt;<BR>
&gt;&gt; So I think that the right course of action is that folks =
doubting ICE<BR>
&gt;&gt; reliability<BR>
&gt;&gt; make their case (by that I mean specific arguments rather =
than<BR>
emotional<BR>
&gt;&gt; debates<BR>
&gt;&gt; about red-colored nails and other hardware).<BR>
&gt;<BR>
&gt;Er, my &quot;specific argument&quot; is &quot;until it's been proven =
to work, it<BR>
hasn't<BR>
&gt;been proven to work&quot;.&nbsp; It's tautological, but no less true =
for it.<BR>
&gt;<BR>
&gt;Indeed, if you're *not* doubting ICE, why aren't you?<BR>
<BR>
Hi David,<BR>
<BR>
I'm not entirely sure what is the step you are suggesting?<BR>
<BR>
I think ICE is architecturally right answer to the right problem.<BR>
Notwithstanding that, there are certainly some things I prefer<BR>
differently (e.g., the infamous debate about proactive hole =
punching)<BR>
or may turn out to be bugs. I think though that the key problem at<BR>
this point is of institutional nature. In the IETF language, =
&quot;hasn't<BR>
been proven to work&quot; is adequate to &quot;lacking running =
code&quot; and this<BR>
appears to be a phenomen which is indeed occuring in the IETF and<BR>
is not unique to ICE.<BR>
<BR>
The question is how to get this fixed?<BR>
<BR>
Some other standardization bodies enjoying the amount of =
contributions<BR>
and complexity as the IETF is enjoying now (S in SIP does not really<BR>
stand for Simple), proceed with 'after-fact filters', in that they<BR>
produce the initial specs faster and later produce 'profiles' which<BR>
separate what has proved and what less. Plus -bis versions are =
produced<BR>
for whatever appears worth puttting effort in it (i.e., for which =
there<BR>
is running code and field evidence of improvement need).This could =
be<BR>
an approach&nbsp; which may be more feasible in today's IETF's dimension =
than<BR>
<BR>
trying to get back to historical roots.<BR>
<BR>
-jiri<BR>
<BR>
<BR>
<BR>
_______________________________________________<BR>
Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org=
/mailman/listinfo/sip</A><BR>
This list is for NEW development of the core SIP Protocol<BR>
Use sip-implementors@cs.columbia.edu for questions on current sip<BR>
Use sipping@ietf.org for new developments on the application of sip<BR>
<BR>
<BR>
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From sip-bounces@ietf.org Sat Jul 21 21:40:42 2007
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Date: Sat, 21 Jul 2007 21:40:01 -0400
Subject: Re: [Sip] Re: AW: [Sipping] comments on draft-burger-sip-info-00
From: Eric Burger <eburger@bea.com>
To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
	"Jesske, R" <R.Jesske@t-com.net>
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No, the point is you almost always use more messages by NOT using INFO.
However, fewer messages and not interoperable is not as good as more
messages and interoperable.


On 7/21/07 2:51 PM, "Christer Holmberg (JO/LMF)"
<christer.holmberg@ericsson.com> wrote:

>=20
>=20
> Hi,
>=20
>> [Moved to SIP by chair request]
>>=20
>> MUMBLEBRATZ for UA-to-UA communication will always work if
>> you own the endpoints.  Might as well use INFO for MUMBLEBRATZ.
>>=20
>> If one wants interoperable communication or if you want to
>> deploy equipment from more than one vendor, or at least not
>> having to specify exact behavior (i.e., creating your own
>> proprietary protocol), INFO doesn't work, as already documented.
>=20
> If you are going to use INFO to carry videoControl messages, of course th=
ose
> messages, and the meaning of those, must be defined and documented - in o=
rder
> to make them non-proprietary.
>=20
> It's the same with ISUP. You use INFO to carry SPECIFIED ISUP messages, a=
nd
> endpoints supporting ISUP know the procedures how to deal with the messag=
es,
> because it is defined in standard documentation.
>=20
>> For that matter, we can save a ton of messages if we drop all
>> the pesky negotiation in SIP.  Why bother if we just "know"
>> what the endpoints will accept.  We just need to start
>> sending RTP and things will just work :-)
>=20
> Please show me message you save by not using INFO. The negotiation can in=
 most
> cases be "encapsulated" in SIP messages you are going to send anyway.
>=20
> But, if you use an event based mechanism, you actually WILL get more SIP
> messages, because you need to send the NOTIFY request(s).
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>>=20
>>=20
>> On 7/18/07 10:10 AM, "Jesske, R" <R.Jesske@t-com.net> wrote:
>>=20
>>> Dear all,
>>> I'm concerned about the discussion here and the aim of the draft to
>>> restrict the use of INFO only to SIP-T and Q-SIG.
>>>=20
>>> If you now change restrict the RFC2976 you will tear down each
>>> implementation already done based on RFC2976. Like for DTMF that is
>>> broadly used within the world. So you are against backward
>> compatibility?
>>>=20
>>> The draft is claiming that INFO has to be routed over each
>> proxy where
>>> the INVITE is routed to. I do not know what's the problem
>> with that. A
>>> proxy forwards and the question is how much load will this
>> cause. UA's
>>> can sent re-INVITES or REGISTER, RE-REGISTER a couple of
>> times and nobody claims that.
>>> INFO is used for UA to UA communication so is this not
>> proper enough
>>> that at least the UA's will know what to do with the MIME?
>>> INFO is ideal to sent information that has to correlate with the
>>> Dialog using other mechanisms like an 2nd stream like http or a
>>> MESSAGE or a other control protocol can not fulfil this.
>> Even it could
>>> be get lost when you want to traverse several SIP domains
>> or if you interwork with the PSTN/ISDN.
>>>=20
>>> The Example of DTMF for INFO can only be done at mid dialog.
>>>=20
>>> Using KPML --> We have minimum 4 more Messages. No additional Load?
>>>=20
>>>=20
>>>=20
>>> Best Regards
>>>=20
>>> Roland
>>>=20
>>>=20
>>>=20
>>>> -----Urspr=FCngliche Nachricht-----
>>>> Von: Christer Holmberg (JO/LMF)
>>>> [mailto:christer.holmberg@ericsson.com]
>>>> Gesendet: Mittwoch, 18. Juli 2007 12:18
>>>> An: Dean Willis
>>>> Cc: IETF Sipping List
>>>> Betreff: RE: [Sipping] comments on draft-burger-sip-info-00
>>>>=20
>>>>=20
>>>>=20
>>>> Hi,
>>>>=20
>>>>>>>> - I disagree with the statement in chapter 1 regarding
>>>> the "context
>>>>>>>> for interpreting". The context is associated with the
>>>> content type.
>>>>>>>>=20
>>>>>>>=20
>>>>>>> Content type is, as I have repeatedly argued, completely
>>>> insufficient
>>>>=20
>>>>>>> for interpretation of context in the general case. What,
>>>> for example,
>>>>=20
>>>>>>> does it mean when an INFO shows up with a wav file
>>>> attached? Play it?
>>>>>>> Record it? Parse it using speech recognition and pass it
>>>> to a command
>>>>=20
>>>>>>> processor?
>>>>>>=20
>>>>>> The dialog (you can't send INFOs out-of-dialog, as far as I
>>>> know) will
>>>>=20
>>>>>> be associated with an application, and if the application
>>>> doesn't know
>>>>=20
>>>>>> what to do with it, it can promt the user: "You have now
>> received a
>>>>>> file. What do you want to do with it? Save it? Play it?
>> Display it?
>>>>>> Delete it?".
>>>>>=20
>>>>> And how does the receiver know what the sender intended?
>>>>>=20
>>>>> Assume you're taking to my "phone" application. You send digits.
>>>>> Should those be rendered to me, or fed into the user
>> interface of a
>>>>> phone-based game we happen to be playing?
>>>>=20
>>>> How do you solve it with 2833?
>>>>=20
>>>> How do you solve it with KPML?
>>>>=20
>>>> You could even have the same problem with normal voice, e.g.
>>>> if the game
>>>> is controlled using voice recognition. Shall the received voice be
>>>> fed to the speaker, or fed to the game application?
>>>>=20
>>>> In some cases it may not matter if the data is fed to multiple
>>>> applications. But, if you do need to dispatch data so specific
>>>> applications you either need some kind of application
>> reference, or
>>>> you handle the dispatching locally in your device (the game
>>>> application can tell it wants DTMFs, and/or the device chooses -
>>>> either automatically or manually - which application will get the
>>>> data etc). Or you use different SIP sessions.
>>>>=20
>>>> In other cases you will not have the problem. For example,
>>>> videoControl messages would normally be fed to a video control
>>>> application, and nowhere else.
>>>>=20
>>>> Regards,
>>>>=20
>>>> Christer
>>>>=20
>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> Sipping mailing list
>> https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list is for NEW development of the application of SIP Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>> sip@ietf.org for new developments of core SIP
>>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP Use
>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>> sip@ietf.org for new developments of core SIP
>>=20
>>=20
>> Notice:  This email message, together with any attachments,
>> may contain information  of  BEA Systems,  Inc.,  its
>> subsidiaries  and  affiliated entities,  that may be
>> confidential,  proprietary,  copyrighted  and/or legally
>> privileged, and is intended solely for the use of the
>> individual or entity named in this message. If you are not
>> the intended recipient, and have received this message in
>> error, please immediately return this by email and then delete it.
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>=20
>=20



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ved this message in error, please immediately return this by email and then=
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From sip-bounces@ietf.org Sat Jul 21 21:41:35 2007
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Date: Sat, 21 Jul 2007 21:41:18 -0400
Subject: Re: [Sip] INFO and Content Negotiation
From: Eric Burger <eburger@bea.com>
To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
	IETF SIP <sip@ietf.org>
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The point is that content type alone is not sufficient for INFO negotiation=
.
You are a RFC 4722-aware device.  You advertise you accept a DTMF payload
type.  I am an INFO-aware device.  I see you understand a DTMF payload, so =
I
send it to you in an *INFO* message.  You barf, because you were expecting
KPML.


On 7/21/07 2:51 PM, "Christer Holmberg (JO/LMF)"
<christer.holmberg@ericsson.com> wrote:

>=20
>=20
> Hi,
>=20
> I don't really understand your issue here... Did you send the text using =
INFO,
> or something? ;)
>=20
> Regards,
>=20
> Christer
>=20
>> -----Original Message-----
>> From: Eric Burger [mailto:eburger@bea.com]
>> Sent: 20. hein=E4kuuta 2007 23:14
>> To: IETF SIP
>> Subject: [Sip] INFO and Content Negotiation
>>=20
>> 12 second proof point of why content type doesn't work.
>>=20
>> So here comes an INVITE.  I am an INFO-friendly UAS.  I
>> negotiate by using Accept.
>>=20
>> This INVITE has the line:
>> Accept: application/kpml-response+xml
>>=20
>>=20
>> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?
>> Because it accepts a DTMF content type.  Excellent - I can
>> now send them DTMF.
>>=20
>> The user enters a digit, and I send an INFO with type
>> application/kpml-response+xml.
>>=20
>> UAC says, "What the heck was that?"
>>=20
>>=20
>> Notice:  This email message, together with any attachments,
>> may contain information  of  BEA Systems,  Inc.,  its
>> subsidiaries  and  affiliated entities,  that may be
>> confidential,  proprietary,  copyrighted  and/or legally
>> privileged, and is intended solely for the use of the
>> individual or entity named in this message. If you are not
>> the intended recipient, and have received this message in
>> error, please immediately return this by email and then delete it.
>>=20
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>=20
>=20



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rivileged, and is intended solely for the use of the individual or entity n=
amed in this message. If you are not the intended recipient, and have recei=
ved this message in error, please immediately return this by email and then=
 delete it.


_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Sat Jul 21 21:45:13 2007
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1)	IANA registry.

The current IANA registry contents created by RFC 4412 is merely:

  Header Name        compact    Reference
  -----------------  -------    ---------
  Accept-Resource-Priority      [RFC4412]
  Resource-Priority             [RFC4412]

The only IANA registry change required to to add this draft to the
reference against Resource-Priority, and not to do what is currently
listed in section 4.

Additionally statements in section 1:

   This document IANA registers the Resource-Priority header for usage=20
   in responses.

And

   The above is to replace what is currently stated by RFC 4412, and=20
   what is IANA registered.

Should also be corrected.

2)	Section 3.1 of RFC 4412 specifies:

   There is no protocol requirement that all requests within a SIP
   dialog or session use the 'Resource-Priority' header field.  Local
   administrative policy MAY mandate the inclusion of the
   'Resource-Priority' header field in all requests.  Implementations of
   this specification MUST allow inclusion to be either by explicit user
   request or automatic for all requests.

I assume that the usage of the header in responses is an extension of
the administrative policy stated above, and therefore this linkage needs
to be made. For example, I read the above text to indicate that a
stateful proxy could receive the header in a initial INVITE request, and
treat all requests and responses in the same dialog with priority. In
this case, I therefore do not need the header in subsequent requests,
and equally I don't need it in any responses either.

3)	RFC 4412 divides the SIP methods into a set of classes giving
various levels of support. I assume that these classes also apply for
the support of responses, but I can currently find no text that
indicates that.

4)	RFC 4412 indicates for example:

   A SIP request with a 'Resource-Priority' indication can be treated
   differently in these situations:

   1.  The request can be given elevated priority for access to PSTN
       gateway resources, such as trunk circuits.

   2.  The request can interrupt lower-priority requests at a user
       terminal, such as an IP phone.

   3.  The request can carry information from one multi-level priority
       domain in the telephone network (e.g., using the facilities of
       Q.735.3 [Q.735.3]) to another, without the SIP proxies themselves
       inspecting or modifying the header field.

   4.  In SIP proxies and back-to-back user agents, requests of higher
       priorities may displace existing signaling requests or bypass
       PSTN gateway capacity limits in effect for lower priorities.

This also applies equally for the responses discussed in this document.
We probably need some general text that basically says that what RFC
4412 specifies for requests, also applies to responses when implementing
this document. This is both at the requirement level and the discussion
level.

5)	Section 2.3 of this document indicates:

   SIP Proxies MAY process the Resource-Priority header in responses;=20
   meaning, in certain environments, the choice of whether or not to=20
   process the RPH will not be in doubt. =20

Section 4.7.3 of RFC 4412 states:

   SIP proxies MAY ignore the 'Resource-Priority' header field.  SIP
   proxies MAY reject any unauthenticated request bearing that header
   field.

However, I assume that what 2.3 is attempting to state is not the above
for responses, but rather: Even though the proxy processes the header in
requests, it may ignore the header in responses.



Regards

Keith


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Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
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This is the way MIME works.  Please don't change it!


On 7/18/07 9:44 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

[snip] 
> For multipart/alternative - I don't understand the requirement to order
> them from least to most 'rich'. Is that a general MIME behavior or are
> we defining it that way? If we want to prioritize, shouldn't we use an
> explicit q-value or something? Using order seems like it will be brittle
> (as it has been in SDP where m-lines were never numbered).
[snip]


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Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
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Seems like a good idea.  However, I would not go so far as to mandate
iterative evaluation.  I would offer that the UA evaluates as needed.


On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:

> Hi,
> I suggest Content Indirection Mechanism defined in rfc4483 should also be
> considered in this draft.
> 
> It is a little special, an example from rfc4483:
> 6.1.  Single Content Indirection
> 
>            INVITE sip:boromir@example.com SIP/2.0
>            Accept: message/external-body application/sdp
>            Content-Type: message/external-body;
>                 ACCESS-TYPE=URL;
>                 URL="http://www.example.net/party/06/2002/announcement";
>                 EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
>                 size=231
>            Content-Length: 105
> 
>            Content-Type: application/sdp
>            Content-Disposition: session
>            Content-ID: <4e5562cd1214427d@example.net>
> 
> In addition, what happens to the INVITE request if the recipient can not get
> the indirect content?
> 
> Nits: a letter "l" is missing in the title.
> Internet-Draft         Message Body Hand"l"ing in SIP              May 2007
> 
> Cheers,
> Qian
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, July 20, 2007 2:30 AM
> To: Gonzalo Camarillo
> Cc: IETF SIP List
> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
> 
> Just one comment in addition to what Gonzalo said:
> 
> Gonzalo Camarillo wrote:
> 
>>> The use case described for multipart/alternative - new SDP mechanisms
>>> - seems kind of bogus. We've decided not to pursue that. Do we have a
>>> REAL use case for this?
>> 
>> AFAICT, we have decided not to pursue SDPng but we still need a
>> mechanism in order to be able to, at some point, migrate to new session
>> description formats. The current consensus, as I understand it, is that
>> multipart/alternative should not be used to provide alternative SDP
>> descriptions, but that it could be used to provide alternative session
>> descriptions written in different formats.
> 
> The other classic example, because it is just like email, is MESSAGE
> with a multipart/alternative containing plain text and html.
> 
> Paul
> 
> 
> _______________________________________________
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> 
> 
> 
> 
> _______________________________________________
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Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides
	for body (MIME) handling)
From: Eric Burger <eburger@bea.com>
To: Paul Kyzivat <pkyzivat@cisco.com>,
	Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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I would offer this is where we can "add value" to MIME for SIP usage.  Note
that I do not say, "deviate from MIME."  I would suggest we NOT try to
deviate from MIME in any respect.


On 7/19/07 9:59 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
[snip] 
> I think we just have to grandfather the use of 'refer' in SIP, and
> document that the usage in sip does not mean "render", but rather means
> that the usage must be determined by other context - specifically the
> type of message or response and the content type. IOW, 'render' in sip
> can be thought of as a peculiar spelling for 'by-context'. It might be
> helpful to give guidance for development of future specs about when
> 'render' should be used vs defining a new disposition.
[snip]


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From sip-bounces@ietf.org Sat Jul 21 21:51:55 2007
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Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides
	for body (MIME) handling)
From: Eric Burger <eburger@bea.com>
To: Qian Sun <sunqian@huawei.com>,
	Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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I respectfully disagree on this one.  Here we are not talking about
disposition, but about the context of the content.  In the case of
"advertisement", we are noting the context of the content.  It is up to the
application (user) to decide to display ads or block ads.  It is NOT up to
the sender.


On 7/20/07 6:04 AM, "Qian Sun" <sunqian@huawei.com> wrote:

> Hi,
> 
> If we define a new Content Disposition type "advertisement" so that the
> operator or service provider may insert advertisement in the all kinds of
> message body. And we want the advertisement content 'render' to the users.
> Could we use multiple content dispositions ("advertisement" and 'render') for
> a single content part? Or only define a new Content Disposition Parameters
> "advertisement" instead?
> 
> This is a real requirement, OMA and GSM is developing the mobile advertisement
> spec. the current advertisement way in Email or Webpage is very "evil", it is
> difficult for the user's client to distinguish the real content from
> advertisement, sometimes this is very disgusting or confusing. A new Content
> Disposition type "advertisement" or something else could be helpful, the
> advertisement may be displayed in a place different from the content.
> 
> Cheers,
> Qian
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, July 20, 2007 2:15 AM
> To: Gonzalo Camarillo
> Cc: sip
> Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides for
> body (MIME) handling)
> 
> 
> 
> Gonzalo Camarillo wrote:
>> Hi,
>> 
>> yes, I agree there is a bigger issue with 'render'. Even though this
>> bigger issue does not directly relate to multipart, we can also tackle
>> it in the same draft and clarify how 'render' is used and what it means.
> 
> Sounds good. Maybe its just a matter of changing the title of the draft
> to match the scope. Something like:
> draft-sipping-content-type-disp-clarification-nn.txt
> 
> Paul
> 
>> Cheers,
>> 
>> Gonzalo
>> 
>> 
>> Paul Kyzivat wrote:
>>> 
>>> 
>>> Gonzalo Camarillo wrote:
>>>> Hi Paul,
>>>> 
>>>>> Regarding Content-Disposition in multiparts:
>>>>> 
>>>>> While it seems "wrong" to me, I can live with using "render" as the
>>>>> normal disposition for multipart, *if* we can do a careful job of
>>>>> defining what "render" actually means for SIP. Used this way it
>>>>> clearly doesn't always mean to display the content to a user.
>>>> 
>>>> An entity receiving a 'multipart/mixed' body will process its body
>>>> parts based on their disposition types.If the disposition type of a
>>>> particular body part is not understood, it will be displayed to the
>>>> user because the default disposition type is 'render'.
>>> 
>>> There is a bigger problem with 'render' than just multipart.
>>> 
>>> We have many uses of body parts that never mention Content-Disposition
>>> at all. The related examples don't use it, and so they all get a
>>> default of 'render'. This is particularly the case with events. E.g.
>>> 
>>> - the notification for REFER contains type application/sipfrag.
>>>   Its probably not intended to be "rendered" in any literal sense,
>>>   though some conclusion about success or failure may be.
>>> 
>>> - usage of filter bodies in presence subscriptions (RFC 4660)
>>>   doesn't have a specified disposition, so it implicitly uses render.
>>>   Its hard for me to convince myself that what the presence server
>>>   does with these can be described as rendering.
>>> 
>>> - In fact, the only place I can think of where 'render' is (implicitly)
>>>   used in sip and has the appropriate connotation is in MESSAGE.
>>> 
>>> I think we just have to grandfather the use of 'refer' in SIP, and
>>> document that the usage in sip does not mean "render", but rather
>>> means that the usage must be determined by other context -
>>> specifically the type of message or response and the content type.
>>> IOW, 'render' in sip can be thought of as a peculiar spelling for
>>> 'by-context'. It might be helpful to give guidance for development of
>>> future specs about when 'render' should be used vs defining a new
>>> disposition.
>>> 
>>>> Therefore, the 'multipart/mixed' body will never be processed based
>>>> on its disposition type.
>>> 
>>> Based on my discussion above, I would consider this a case of the
>>> multipart being processed according to its context *because* its
>>> content-disposition is 'refer'. If it had some other
>>> content-disposition then that might mean that it should be processed
>>> specially. E.g. if a multipart/mixed had disposition 'by-reference'
>>> then that needs to be taken into account, and the processing of the
>>> multipart becomes conditional on the reference.
>>> 
>>> Also, there are only certain contexts in which multipart should be
>>> processed "normally". Consider a NOTIFY from a REFER, with a
>>> Content-Type of message/sipfrag, which happened to include a body as
>>> part of the fragment. (This is legal.) That body might contain a
>>> multipart/mixed, and that might contain other stuff. But I think the
>>> processing of all of that is dependent on processing of the sipfrag.
>>> For instance, if something inside that had handling=required, I don't
>>> think that means that processing of the NOTIFY fails because the
>>> recipient can't handle that.
>>> 
>>>> This means that it does not make much sense to talk about its
>>>> disposition type as a whole.
>>> 
>>> Counterexample above.
>>> 
>>>> However, we need to have one in order to be able to set its handling
>>>> parameter. That's why we can either keep 'render' explaining that it
>>>> is actually never used or define a new disposition type.
>>> 
>>> I can go either way on that. But because there is existing usage, and
>>> because 'render' semantics are already so messed up, I think it makes
>>> more sense to stick with 'render' and just clarify its meaning.
>>> 
>>>> Since existing implementations will not understand our newly defined
>>>> disposition type, they will treat it as 'render' anyway... but the
>>>> resulting behavior would be the same. That's why it may make sense to
>>>> keep using 'render'.
>>> 
>>> Yup.
>>> 
>>>> On the other hand, semantically, it would make more sense to define a
>>>> new disposition type.
>>>> 
>>>> Also, if we decide to define the 'by-reference' disposition type (per
>>>> a different email thread), it could be used here as well.
>>> 
>>> It could be used when there is actually a reference to it. But it
>>> would not be an appropriate disposition to use when there is no
>>> reference to the multipart.
>>> 
>>>     Thanks,
>>>     Paul
>>> 
>>>> Cheers,
>>>> 
>>>> Gonzalo
>>>> 
>>>> 
>>>> 
>>>>> Gonzalo Camarillo wrote:
>>>>>> Folks,
>>>>>> 
>>>>>> here you have the slides I have put together for the body (MIME)
>>>>>> handling presentation on Monday:
>>>>>> 
>>>>>> http://users.piuha.net/gonzalo/temp/ietf69-sip-camarillo-body-handling.pp
>>>>>> t 
>>>>>> 
>>>>>> 
>>>>>> These slides relate to this draft:
>>>>>> 
>>>>>> http://www.ietf.org/internet-drafts/draft-camarillo-sip-body-handling-01.
>>>>>> txt 
>>>>>> 
>>>>>> 
>>>>>> The slides discuss all the open issues that have been brought up in
>>>>>> the list in the last months.
>>>>>> 
>>>>>> Comments are welcome.
>>>>>> 
>>>>>> Thanks,
>>>>>> 
>>>>>> Gonzalo
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol
>>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>>> 
>>>> 
>> 
> 
> 
> _______________________________________________
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> 
> 
> 
> 
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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1)	I am not convinced this document does updated RFC 4412. RFC 4412
creates the namespace. This document merely adds new items to the IANA
registry created by that document. The document header should be
modified accordingly, and the following text in the abstract also
corrected:

   This=20
   document intends to update RFC 4412, as a Proposed Standard document
   if published by the RFC-Editor.

2)	I believe the abstract could usefully summarise the new
namespaces that are created (in the same way as most drafts that define
new headers identify them in the abstract).

3)	In section 3, I do not see the need to include the following
text in the document, as it is already in RFC 4412, for which all the
requirements apply, not just this one.

   As stated in Section 9 of RFC 4412 [RFC4412], an IANA registered=20
   namespace SHOULD NOT change the number, and MUST NOT change the=20
   relative priority order, of its assigned priority-values.

4)	NIT. Tag all mentions of RFC4412 as references.


Regards

Keith


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Eric Burger wrote:
> I would offer this is where we can "add value" to MIME for SIP usage.  Note
> that I do not say, "deviate from MIME."  I would suggest we NOT try to
> deviate from MIME in any respect.

Oh MIME sage, your comment is inscrutable! :-)

I don't understand in this case where the line is between "adding value" 
and "deviating".

If we try to ascribe some common meaning to "render" then I believe sip 
is already violating it. If we have thus already "deviated" from MIME, 
which takes precedence: correcting the deviation, or backward compatibility.

My assumption is that backward compatibility is more important.

	Paul

> On 7/19/07 9:59 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> [snip] 
>> I think we just have to grandfather the use of 'refer' in SIP, and
>> document that the usage in sip does not mean "render", but rather means
>> that the usage must be determined by other context - specifically the
>> type of message or response and the content type. IOW, 'render' in sip
>> can be thought of as a peculiar spelling for 'by-context'. It might be
>> helpful to give guidance for development of future specs about when
>> 'render' should be used vs defining a new disposition.
> [snip]
> 
> 
> Notice:  This email message, together with any attachments, may contain information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entities,  that may be confidential,  proprietary,  copyrighted  and/or legally privileged, and is intended solely for the use of the individual or entity named in this message. If you are not the intended recipient, and have received this message in error, please immediately return this by email and then delete it.
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> 


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One of the tasks of the draft is to make support for at least some 
subtypes of multipart (e.g. mixed) mandatory.

I hope this is not a suggestion to make dereferencing of external bodies 
mandatory.

If the request is just to include some clarification of the use and 
handling of external bodies in this draft then I guess that does sound 
useful.

	Paul

Eric Burger wrote:
> Seems like a good idea.  However, I would not go so far as to mandate
> iterative evaluation.  I would offer that the UA evaluates as needed.
> 
> 
> On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:
> 
>> Hi,
>> I suggest Content Indirection Mechanism defined in rfc4483 should also be
>> considered in this draft.
>>
>> It is a little special, an example from rfc4483:
>> 6.1.  Single Content Indirection
>>
>>            INVITE sip:boromir@example.com SIP/2.0
>>            Accept: message/external-body application/sdp
>>            Content-Type: message/external-body;
>>                 ACCESS-TYPE=URL;
>>                 URL="http://www.example.net/party/06/2002/announcement";
>>                 EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
>>                 size=231
>>            Content-Length: 105
>>
>>            Content-Type: application/sdp
>>            Content-Disposition: session
>>            Content-ID: <4e5562cd1214427d@example.net>
>>
>> In addition, what happens to the INVITE request if the recipient can not get
>> the indirect content?
>>
>> Nits: a letter "l" is missing in the title.
>> Internet-Draft         Message Body Hand"l"ing in SIP              May 2007
>>
>> Cheers,
>> Qian
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Friday, July 20, 2007 2:30 AM
>> To: Gonzalo Camarillo
>> Cc: IETF SIP List
>> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
>>
>> Just one comment in addition to what Gonzalo said:
>>
>> Gonzalo Camarillo wrote:
>>
>>>> The use case described for multipart/alternative - new SDP mechanisms
>>>> - seems kind of bogus. We've decided not to pursue that. Do we have a
>>>> REAL use case for this?
>>> AFAICT, we have decided not to pursue SDPng but we still need a
>>> mechanism in order to be able to, at some point, migrate to new session
>>> description formats. The current consensus, as I understand it, is that
>>> multipart/alternative should not be used to provide alternative SDP
>>> descriptions, but that it could be used to provide alternative session
>>> descriptions written in different formats.
>> The other classic example, because it is just like email, is MESSAGE
>> with a multipart/alternative containing plain text and html.
>>
>> Paul
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> Notice:  This email message, together with any attachments, may contain information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entities,  that may be confidential,  proprietary,  copyrighted  and/or legally privileged, and is intended solely for the use of the individual or entity named in this message. If you are not the intended recipient, and have received this message in error, please immediately return this by email and then delete it.
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> 


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Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides
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From: Eric Burger <eburger@bea.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Well, we are only sort-of munging MIME.  The default is always RENDER.
However, *what* is a render is an exercise for the protocol to specify.
That is the wiggle room for SIP to say "render means by-context."

What we do not have the wiggle room for is to say the default
content-disposition is something other than RENDER.


On 7/22/07 12:17 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> 
> 
> 
> Eric Burger wrote:
>> I would offer this is where we can "add value" to MIME for SIP usage.  Note
>> that I do not say, "deviate from MIME."  I would suggest we NOT try to
>> deviate from MIME in any respect.
> 
> Oh MIME sage, your comment is inscrutable! :-)
> 
> I don't understand in this case where the line is between "adding value"
> and "deviating".
> 
> If we try to ascribe some common meaning to "render" then I believe sip
> is already violating it. If we have thus already "deviated" from MIME,
> which takes precedence: correcting the deviation, or backward compatibility.
> 
> My assumption is that backward compatibility is more important.
> 
>         Paul
> 
>> On 7/19/07 9:59 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
>> [snip]
>>> I think we just have to grandfather the use of 'refer' in SIP, and
>>> document that the usage in sip does not mean "render", but rather means
>>> that the usage must be determined by other context - specifically the
>>> type of message or response and the content type. IOW, 'render' in sip
>>> can be thought of as a peculiar spelling for 'by-context'. It might be
>>> helpful to give guidance for development of future specs about when
>>> 'render' should be used vs defining a new disposition.
>> [snip]
>> 
>> 
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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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Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
From: Eric Burger <eburger@bea.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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The latter: be aware of indirection, but (1) not mandatory and in fact (2)
don't bother to fetch the bits unless you absolutely have to.

IMHO.  Just one vote.


On 7/22/07 12:21 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> One of the tasks of the draft is to make support for at least some
> subtypes of multipart (e.g. mixed) mandatory.
> 
> I hope this is not a suggestion to make dereferencing of external bodies
> mandatory.
> 
> If the request is just to include some clarification of the use and
> handling of external bodies in this draft then I guess that does sound
> useful.
> 
>         Paul
> 
> Eric Burger wrote:
>> Seems like a good idea.  However, I would not go so far as to mandate
>> iterative evaluation.  I would offer that the UA evaluates as needed.
>> 
>> 
>> On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:
>> 
>>> Hi,
>>> I suggest Content Indirection Mechanism defined in rfc4483 should also be
>>> considered in this draft.
>>> 
>>> It is a little special, an example from rfc4483:
>>> 6.1.  Single Content Indirection
>>> 
>>>            INVITE sip:boromir@example.com SIP/2.0
>>>            Accept: message/external-body application/sdp
>>>            Content-Type: message/external-body;
>>>                 ACCESS-TYPE=URL;
>>>                 URL="http://www.example.net/party/06/2002/announcement";
>>>                 EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
>>>                 size=231
>>>            Content-Length: 105
>>> 
>>>            Content-Type: application/sdp
>>>            Content-Disposition: session
>>>            Content-ID: <4e5562cd1214427d@example.net>
>>> 
>>> In addition, what happens to the INVITE request if the recipient can not get
>>> the indirect content?
>>> 
>>> Nits: a letter "l" is missing in the title.
>>> Internet-Draft         Message Body Hand"l"ing in SIP              May 2007
>>> 
>>> Cheers,
>>> Qian
>>> 
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: Friday, July 20, 2007 2:30 AM
>>> To: Gonzalo Camarillo
>>> Cc: IETF SIP List
>>> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
>>> 
>>> Just one comment in addition to what Gonzalo said:
>>> 
>>> Gonzalo Camarillo wrote:
>>> 
>>>>> The use case described for multipart/alternative - new SDP mechanisms
>>>>> - seems kind of bogus. We've decided not to pursue that. Do we have a
>>>>> REAL use case for this?
>>>> AFAICT, we have decided not to pursue SDPng but we still need a
>>>> mechanism in order to be able to, at some point, migrate to new session
>>>> description formats. The current consensus, as I understand it, is that
>>>> multipart/alternative should not be used to provide alternative SDP
>>>> descriptions, but that it could be used to provide alternative session
>>>> descriptions written in different formats.
>>> The other classic example, because it is just like email, is MESSAGE
>>> with a multipart/alternative containing plain text and html.
>>> 
>>> Paul
>>> 
>>> 
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
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>> 
>> 
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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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On Jul 20, 2007, at 4:08 PM, Francois Audet wrote:
>
> The last sentence of section 8 says "The SIP keepalive STUN usage
> requires no backwards
> compatibility.".
>
> It's not clear "with what".
>
> I'm assuming that you mean it works with rfc3489bis but not RFC 3489?

correct.  this is one of the things listed that a usage of stun needs  
to handle. i'll put this in the document.

thanks,
-rohan

> Please clarify (in the document and on the mailing list).



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Subject: RE: [Sip] Re: AW: [Sipping] comments on draft-burger-sip-info-00
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Hi,=20

>No, the point is you almost always use more messages by NOT using INFO.

I think issues like that should also be documented in your draft.

>However, fewer messages and not interoperable is not as good=20
>as more messages and interoperable.

Well, the initeroperability issue is still under discussion.

Regards,

Christer







> > Hi,
> >=20
> >> [Moved to SIP by chair request]
> >>=20
> >> MUMBLEBRATZ for UA-to-UA communication will always work if you own=20
> >> the endpoints.  Might as well use INFO for MUMBLEBRATZ.
> >>=20
> >> If one wants interoperable communication or if you want to deploy=20
> >> equipment from more than one vendor, or at least not having to=20
> >> specify exact behavior (i.e., creating your own proprietary=20
> >> protocol), INFO doesn't work, as already documented.
> >=20
> > If you are going to use INFO to carry videoControl=20
> messages, of course=20
> > those messages, and the meaning of those, must be defined and=20
> > documented - in order to make them non-proprietary.
> >=20
> > It's the same with ISUP. You use INFO to carry SPECIFIED ISUP=20
> > messages, and endpoints supporting ISUP know the procedures how to=20
> > deal with the messages, because it is defined in standard=20
> documentation.
> >=20
> >> For that matter, we can save a ton of messages if we drop all the=20
> >> pesky negotiation in SIP.  Why bother if we just "know"
> >> what the endpoints will accept.  We just need to start sending RTP=20
> >> and things will just work :-)
> >=20
> > Please show me message you save by not using INFO. The=20
> negotiation can=20
> > in most cases be "encapsulated" in SIP messages you are=20
> going to send anyway.
> >=20
> > But, if you use an event based mechanism, you actually WILL=20
> get more=20
> > SIP messages, because you need to send the NOTIFY request(s).
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >>=20
> >>=20
> >> On 7/18/07 10:10 AM, "Jesske, R" <R.Jesske@t-com.net> wrote:
> >>=20
> >>> Dear all,
> >>> I'm concerned about the discussion here and the aim of=20
> the draft to=20
> >>> restrict the use of INFO only to SIP-T and Q-SIG.
> >>>=20
> >>> If you now change restrict the RFC2976 you will tear down each=20
> >>> implementation already done based on RFC2976. Like for=20
> DTMF that is=20
> >>> broadly used within the world. So you are against backward
> >> compatibility?
> >>>=20
> >>> The draft is claiming that INFO has to be routed over each
> >> proxy where
> >>> the INVITE is routed to. I do not know what's the problem
> >> with that. A
> >>> proxy forwards and the question is how much load will this
> >> cause. UA's
> >>> can sent re-INVITES or REGISTER, RE-REGISTER a couple of
> >> times and nobody claims that.
> >>> INFO is used for UA to UA communication so is this not
> >> proper enough
> >>> that at least the UA's will know what to do with the MIME?
> >>> INFO is ideal to sent information that has to correlate with the=20
> >>> Dialog using other mechanisms like an 2nd stream like http or a=20
> >>> MESSAGE or a other control protocol can not fulfil this.
> >> Even it could
> >>> be get lost when you want to traverse several SIP domains
> >> or if you interwork with the PSTN/ISDN.
> >>>=20
> >>> The Example of DTMF for INFO can only be done at mid dialog.
> >>>=20
> >>> Using KPML --> We have minimum 4 more Messages. No=20
> additional Load?
> >>>=20
> >>>=20
> >>>=20
> >>> Best Regards
> >>>=20
> >>> Roland
> >>>=20
> >>>=20
> >>>=20
> >>>> -----Urspr=FCngliche Nachricht-----
> >>>> Von: Christer Holmberg (JO/LMF)
> >>>> [mailto:christer.holmberg@ericsson.com]
> >>>> Gesendet: Mittwoch, 18. Juli 2007 12:18
> >>>> An: Dean Willis
> >>>> Cc: IETF Sipping List
> >>>> Betreff: RE: [Sipping] comments on draft-burger-sip-info-00
> >>>>=20
> >>>>=20
> >>>>=20
> >>>> Hi,
> >>>>=20
> >>>>>>>> - I disagree with the statement in chapter 1 regarding
> >>>> the "context
> >>>>>>>> for interpreting". The context is associated with the
> >>>> content type.
> >>>>>>>>=20
> >>>>>>>=20
> >>>>>>> Content type is, as I have repeatedly argued, completely
> >>>> insufficient
> >>>>=20
> >>>>>>> for interpretation of context in the general case. What,
> >>>> for example,
> >>>>=20
> >>>>>>> does it mean when an INFO shows up with a wav file
> >>>> attached? Play it?
> >>>>>>> Record it? Parse it using speech recognition and pass it
> >>>> to a command
> >>>>=20
> >>>>>>> processor?
> >>>>>>=20
> >>>>>> The dialog (you can't send INFOs out-of-dialog, as far as I
> >>>> know) will
> >>>>=20
> >>>>>> be associated with an application, and if the application
> >>>> doesn't know
> >>>>=20
> >>>>>> what to do with it, it can promt the user: "You have now
> >> received a
> >>>>>> file. What do you want to do with it? Save it? Play it?
> >> Display it?
> >>>>>> Delete it?".
> >>>>>=20
> >>>>> And how does the receiver know what the sender intended?
> >>>>>=20
> >>>>> Assume you're taking to my "phone" application. You send digits.
> >>>>> Should those be rendered to me, or fed into the user
> >> interface of a
> >>>>> phone-based game we happen to be playing?
> >>>>=20
> >>>> How do you solve it with 2833?
> >>>>=20
> >>>> How do you solve it with KPML?
> >>>>=20
> >>>> You could even have the same problem with normal voice, e.g.
> >>>> if the game
> >>>> is controlled using voice recognition. Shall the=20
> received voice be=20
> >>>> fed to the speaker, or fed to the game application?
> >>>>=20
> >>>> In some cases it may not matter if the data is fed to multiple=20
> >>>> applications. But, if you do need to dispatch data so specific=20
> >>>> applications you either need some kind of application
> >> reference, or
> >>>> you handle the dispatching locally in your device (the game=20
> >>>> application can tell it wants DTMFs, and/or the device chooses -=20
> >>>> either automatically or manually - which application=20
> will get the=20
> >>>> data etc). Or you use different SIP sessions.
> >>>>=20
> >>>> In other cases you will not have the problem. For example,=20
> >>>> videoControl messages would normally be fed to a video control=20
> >>>> application, and nowhere else.
> >>>>=20
> >>>> Regards,
> >>>>=20
> >>>> Christer
> >>>>=20
> >>>>=20
> >>>>=20
> >>>> _______________________________________________
> >>>> Sipping mailing list
> >> https://www1.ietf.org/mailman/listinfo/sipping
> >>>> This list is for NEW development of the application of SIP Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sip@ietf.org for new developments of core SIP
> >>>>=20
> >>>=20
> >>>=20
> >>> _______________________________________________
> >>> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >>> This list is for NEW development of the application of SIP Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sip@ietf.org for new developments of core SIP
> >>=20
> >>=20
> >> Notice:  This email message, together with any attachments, may=20
> >> contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and =20
> >> affiliated entities,  that may be confidential,  proprietary, =20
> >> copyrighted  and/or legally privileged, and is intended solely for=20
> >> the use of the individual or entity named in this message.=20
> If you are=20
> >> not the intended recipient, and have received this message=20
> in error,=20
> >> please immediately return this by email and then delete it.
> >>=20
> >>=20
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>=20
> >=20
>=20
>=20
>=20
> Notice:  This email message, together with any attachments,=20
> may contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and  affiliated entities,  that may be=20
> confidential,  proprietary,  copyrighted  and/or legally=20
> privileged, and is intended solely for the use of the=20
> individual or entity named in this message. If you are not=20
> the intended recipient, and have received this message in=20
> error, please immediately return this by email and then delete it.
>=20


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



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Subject: RE: [Sip] INFO and Content Negotiation
Date: Sun, 22 Jul 2007 09:10:18 +0200
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	<C2C82DFE.7D3F%eburger@bea.com>
From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Eric Burger" <eburger@bea.com>,
	"IETF SIP" <sip@ietf.org>
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Why would I be expecting KPML, if I have only indicated that I support =
DTMF-over-INFO?

If I was to expect KPML, I would first need to establish a subscription =
for KPML (or, in our "controlled network", the subscription would be =
established by default, and I would know you are going to send KPML).

Regards,

Christer

=20

> -----Original Message-----
> From: Eric Burger [mailto:eburger@bea.com]=20
> Sent: 22. hein=E4kuuta 2007 4:41
> To: Christer Holmberg (JO/LMF); IETF SIP
> Subject: Re: [Sip] INFO and Content Negotiation
>=20
> The point is that content type alone is not sufficient for=20
> INFO negotiation.
> You are a RFC 4722-aware device.  You advertise you accept a=20
> DTMF payload type.  I am an INFO-aware device.  I see you=20
> understand a DTMF payload, so I send it to you in an *INFO*=20
> message.  You barf, because you were expecting KPML.
>=20
>=20
> On 7/21/07 2:51 PM, "Christer Holmberg (JO/LMF)"
> <christer.holmberg@ericsson.com> wrote:
>=20
> >=20
> >=20
> > Hi,
> >=20
> > I don't really understand your issue here... Did you send the text=20
> > using INFO, or something? ;)
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >> -----Original Message-----
> >> From: Eric Burger [mailto:eburger@bea.com]
> >> Sent: 20. hein=E4kuuta 2007 23:14
> >> To: IETF SIP
> >> Subject: [Sip] INFO and Content Negotiation
> >>=20
> >> 12 second proof point of why content type doesn't work.
> >>=20
> >> So here comes an INVITE.  I am an INFO-friendly UAS.  I=20
> negotiate by=20
> >> using Accept.
> >>=20
> >> This INVITE has the line:
> >> Accept: application/kpml-response+xml
> >>=20
> >>=20
> >> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?
> >> Because it accepts a DTMF content type.  Excellent - I can=20
> now send=20
> >> them DTMF.
> >>=20
> >> The user enters a digit, and I send an INFO with type=20
> >> application/kpml-response+xml.
> >>=20
> >> UAC says, "What the heck was that?"
> >>=20
> >>=20
> >> Notice:  This email message, together with any attachments, may=20
> >> contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and =20
> >> affiliated entities,  that may be confidential,  proprietary, =20
> >> copyrighted  and/or legally privileged, and is intended solely for=20
> >> the use of the individual or entity named in this message.=20
> If you are=20
> >> not the intended recipient, and have received this message=20
> in error,=20
> >> please immediately return this by email and then delete it.
> >>=20
> >>=20
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >>=20
> >=20
>=20
>=20
>=20
> Notice:  This email message, together with any attachments,=20
> may contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and  affiliated entities,  that may be=20
> confidential,  proprietary,  copyrighted  and/or legally=20
> privileged, and is intended solely for the use of the=20
> individual or entity named in this message. If you are not=20
> the intended recipient, and have received this message in=20
> error, please immediately return this by email and then delete it.
>=20


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



From sip-bounces@ietf.org Sun Jul 22 03:14:03 2007
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Subject: Re: [Sip] INFO and Content Negotiation
From: Eric Burger <eburger@bea.com>
To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
	IETF SIP <sip@ietf.org>
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That is the point - people have been saying, "I can negotiate INFO by using
Content-Type."  This is a tiny example where using Content-Type to negotiat=
e
"INFO Packages" falls apart.


On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
<christer.holmberg@ericsson.com> wrote:

>=20
>=20
> Why would I be expecting KPML, if I have only indicated that I support
> DTMF-over-INFO?
>=20
> If I was to expect KPML, I would first need to establish a subscription f=
or
> KPML (or, in our "controlled network", the subscription would be establis=
hed
> by default, and I would know you are going to send KPML).
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>> -----Original Message-----
>> From: Eric Burger [mailto:eburger@bea.com]
>> Sent: 22. hein=E4kuuta 2007 4:41
>> To: Christer Holmberg (JO/LMF); IETF SIP
>> Subject: Re: [Sip] INFO and Content Negotiation
>>=20
>> The point is that content type alone is not sufficient for
>> INFO negotiation.
>> You are a RFC 4722-aware device.  You advertise you accept a
>> DTMF payload type.  I am an INFO-aware device.  I see you
>> understand a DTMF payload, so I send it to you in an *INFO*
>> message.  You barf, because you were expecting KPML.
>>=20
>>=20
>> On 7/21/07 2:51 PM, "Christer Holmberg (JO/LMF)"
>> <christer.holmberg@ericsson.com> wrote:
>>=20
>>>=20
>>>=20
>>> Hi,
>>>=20
>>> I don't really understand your issue here... Did you send the text
>>> using INFO, or something? ;)
>>>=20
>>> Regards,
>>>=20
>>> Christer
>>>=20
>>>> -----Original Message-----
>>>> From: Eric Burger [mailto:eburger@bea.com]
>>>> Sent: 20. hein=E4kuuta 2007 23:14
>>>> To: IETF SIP
>>>> Subject: [Sip] INFO and Content Negotiation
>>>>=20
>>>> 12 second proof point of why content type doesn't work.
>>>>=20
>>>> So here comes an INVITE.  I am an INFO-friendly UAS.  I
>> negotiate by
>>>> using Accept.
>>>>=20
>>>> This INVITE has the line:
>>>> Accept: application/kpml-response+xml
>>>>=20
>>>>=20
>>>> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?
>>>> Because it accepts a DTMF content type.  Excellent - I can
>> now send
>>>> them DTMF.
>>>>=20
>>>> The user enters a digit, and I send an INFO with type
>>>> application/kpml-response+xml.
>>>>=20
>>>> UAC says, "What the heck was that?"
>>>>=20
>>>>=20
>>>> Notice:  This email message, together with any attachments, may
>>>> contain information  of  BEA Systems,  Inc.,  its
>> subsidiaries  and
>>>> affiliated entities,  that may be confidential,  proprietary,
>>>> copyrighted  and/or legally privileged, and is intended solely for
>>>> the use of the individual or entity named in this message.
>> If you are
>>>> not the intended recipient, and have received this message
>> in error,
>>>> please immediately return this by email and then delete it.
>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>> sipping@ietf.org for new developments on the application of sip
>>>>=20
>>>=20
>>=20
>>=20
>>=20
>> Notice:  This email message, together with any attachments,
>> may contain information  of  BEA Systems,  Inc.,  its
>> subsidiaries  and  affiliated entities,  that may be
>> confidential,  proprietary,  copyrighted  and/or legally
>> privileged, and is intended solely for the use of the
>> individual or entity named in this message. If you are not
>> the intended recipient, and have received this message in
>> error, please immediately return this by email and then delete it.
>>=20
>=20



Notice:  This email message, together with any attachments, may contain inf=
ormation  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entiti=
es,  that may be confidential,  proprietary,  copyrighted  and/or legally p=
rivileged, and is intended solely for the use of the individual or entity n=
amed in this message. If you are not the intended recipient, and have recei=
ved this message in error, please immediately return this by email and then=
 delete it.


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



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Subject: RE: [Sip] INFO and Content Negotiation
Date: Sun, 22 Jul 2007 09:18:35 +0200
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Eric Burger" <eburger@bea.com>,
	"IETF SIP" <sip@ietf.org>
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Hi,

I still don't get it, but let's take it face-to-face at some point :)

Regards,

Christer=20

> -----Original Message-----
> From: Eric Burger [mailto:eburger@bea.com]=20
> Sent: 22. hein=E4kuuta 2007 10:14
> To: Christer Holmberg (JO/LMF); IETF SIP
> Subject: Re: [Sip] INFO and Content Negotiation
>=20
> That is the point - people have been saying, "I can negotiate=20
> INFO by using Content-Type."  This is a tiny example where=20
> using Content-Type to negotiate "INFO Packages" falls apart.
>=20
>=20
> On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
> <christer.holmberg@ericsson.com> wrote:
>=20
> >=20
> >=20
> > Why would I be expecting KPML, if I have only indicated=20
> that I support=20
> > DTMF-over-INFO?
> >=20
> > If I was to expect KPML, I would first need to establish a=20
> > subscription for KPML (or, in our "controlled network", the=20
> > subscription would be established by default, and I would=20
> know you are going to send KPML).
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >> -----Original Message-----
> >> From: Eric Burger [mailto:eburger@bea.com]
> >> Sent: 22. hein=E4kuuta 2007 4:41
> >> To: Christer Holmberg (JO/LMF); IETF SIP
> >> Subject: Re: [Sip] INFO and Content Negotiation
> >>=20
> >> The point is that content type alone is not sufficient for INFO=20
> >> negotiation.
> >> You are a RFC 4722-aware device.  You advertise you accept a DTMF=20
> >> payload type.  I am an INFO-aware device.  I see you understand a=20
> >> DTMF payload, so I send it to you in an *INFO* message.  You barf,=20
> >> because you were expecting KPML.
> >>=20
> >>=20
> >> On 7/21/07 2:51 PM, "Christer Holmberg (JO/LMF)"
> >> <christer.holmberg@ericsson.com> wrote:
> >>=20
> >>>=20
> >>>=20
> >>> Hi,
> >>>=20
> >>> I don't really understand your issue here... Did you send=20
> the text=20
> >>> using INFO, or something? ;)
> >>>=20
> >>> Regards,
> >>>=20
> >>> Christer
> >>>=20
> >>>> -----Original Message-----
> >>>> From: Eric Burger [mailto:eburger@bea.com]
> >>>> Sent: 20. hein=E4kuuta 2007 23:14
> >>>> To: IETF SIP
> >>>> Subject: [Sip] INFO and Content Negotiation
> >>>>=20
> >>>> 12 second proof point of why content type doesn't work.
> >>>>=20
> >>>> So here comes an INVITE.  I am an INFO-friendly UAS.  I
> >> negotiate by
> >>>> using Accept.
> >>>>=20
> >>>> This INVITE has the line:
> >>>> Accept: application/kpml-response+xml
> >>>>=20
> >>>>=20
> >>>> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?
> >>>> Because it accepts a DTMF content type.  Excellent - I can
> >> now send
> >>>> them DTMF.
> >>>>=20
> >>>> The user enters a digit, and I send an INFO with type=20
> >>>> application/kpml-response+xml.
> >>>>=20
> >>>> UAC says, "What the heck was that?"
> >>>>=20
> >>>>=20
> >>>> Notice:  This email message, together with any attachments, may=20
> >>>> contain information  of  BEA Systems,  Inc.,  its
> >> subsidiaries  and
> >>>> affiliated entities,  that may be confidential,  proprietary,=20
> >>>> copyrighted  and/or legally privileged, and is intended=20
> solely for=20
> >>>> the use of the individual or entity named in this message.
> >> If you are
> >>>> not the intended recipient, and have received this message
> >> in error,
> >>>> please immediately return this by email and then delete it.
> >>>>=20
> >>>>=20
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>>=20
> >>>=20
> >>=20
> >>=20
> >>=20
> >> Notice:  This email message, together with any attachments, may=20
> >> contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and =20
> >> affiliated entities,  that may be confidential,  proprietary, =20
> >> copyrighted  and/or legally privileged, and is intended solely for=20
> >> the use of the individual or entity named in this message.=20
> If you are=20
> >> not the intended recipient, and have received this message=20
> in error,=20
> >> please immediately return this by email and then delete it.
> >>=20
> >=20
>=20
>=20
>=20
> Notice:  This email message, together with any attachments,=20
> may contain information  of  BEA Systems,  Inc.,  its=20
> subsidiaries  and  affiliated entities,  that may be=20
> confidential,  proprietary,  copyrighted  and/or legally=20
> privileged, and is intended solely for the use of the=20
> individual or entity named in this message. If you are not=20
> the intended recipient, and have received this message in=20
> error, please immediately return this by email and then delete it.
>=20


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



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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0936453087==--





From sip-bounces@ietf.org Sun Jul 22 09:48:33 2007
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From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
To: "Cullen Jennings" <fluffy@cisco.com>,
	"Rohan Mahy" <rohan@ekabal.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-outbound-10.txt 
Date: Sun, 22 Jul 2007 15:45:24 +0200
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Extra line of requirement text for section 4.2 on registrations:

When the UAC includes a 'reg-id' parameter in a Contact, it MUST include the 
same parameter and value in refreshes and deregistrations for that Contact.
(could be combined with the remark on re-registrations at the end, the 
latter should also mention de-registrations)

This is needed in (or at least greatly simplifies) a deployment where the 
registrar is separate from the edge proxy, in which case the registrar may 
not be able to determine which flow is being refreshed / deregistered (e.g. 
when TCP connections between the edge proxies and the registrar got 
recycled)

You may want to say a few words on the processing of refreshes and 
deregistrations in section 6, as the matching of bindings is now different 
from RFC3261 (it now includes the instance-id & reg-id)

Regards,
Jeroen

> Internet-Drafts@ietf.org wrote:
>>> 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 : Managing Client Initiated Connections in the Session
>>> Initiation Protocol (SIP) Author(s) : C. Jennings, R. Mahy
>>> Filename : draft-ietf-sip-outbound-10.txt
>>> Pages : 42
>>> Date : 2007-7-12
>>>
>>> The Session Initiation Protocol (SIP) allows proxy servers to
>>>   initiate TCP connections and send asynchronous UDP datagrams to
>>>   User Agents in order to deliver requests.  However, many practical
>>>   considerations, such as the existence of firewalls and Network
>>>   Address Translators (NATs), prevent servers from connecting to
>>>   User Agents in this way.  This specification defines behaviors
>>>   for User Agents, registrars and proxy servers that allow requests
>>>   to be delivered on existing connections established by the User
>>>   Agent. It also defines keep alive behaviors needed to keep NAT
>>>   bindings open and specifies the usage of multiple connections.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-10.txt
>>>
>>> To remove yourself from the I-D Announcement list, send a message to
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>>>
>>> Internet-Drafts are also available by anonymous FTP. Login with the
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>>> logging in, type "cd internet-drafts" and then
>>> "get draft-ietf-sip-outbound-10.txt".
>>>
>>> A list of Internet-Drafts directories can be found in
>>> http://www.ietf.org/shadow.html
>>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>>>
>>> Internet-Drafts can also be obtained by e-mail.
>>>
>>> Send a message to:
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>>>
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>>
>>
>>
>>> _______________________________________________
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>>> This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Sun Jul 22 10:13:29 2007
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Subject: RE: [Sip] Slides for IETF#69
Date: Sun, 22 Jul 2007 16:12:54 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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We have now reached the 24 hour deadline for session #1 and the results
are to say the least DISAPPOINTING.

We have slides for the SIPS slot and for the essential corrections slot
at:

https://datatracker.ietf.org/meeting/69/materials.html

We have apparently not yet received session#1 slides for:

	outbound (Rohan)
	resource priority header issues (James)
	Delivering R-URI and Parameters to UA (Jonathan)
	MIME Body Handling (Gonzalo)

There were some slides on the MIME body handling being circulated by
Gonzalo - are those the ones you intend to use Gonzalo, or is there a
replacement set?

So please get these slides to us as soon as possible - that is assuming
you want to use slides in the session!

The deadline for session#2 slides in 9:00 Monday morning.

Regards

Keith
=09

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: Thursday, July 19, 2007 9:09 PM
> To: sip@ietf.org
> Subject: [Sip] Slides for IETF#69
>=20
> (as WG chair)
>=20
> To all presenters on Monday and Tuesday.
>=20
> If you are presenting slides in the SIP sessions, we would=20
> like these posted to the meeting presentations site well=20
> before the meeting session begins.
>=20
> That means you need to get them to the chairs at least 24=20
> hours in advance of session start (and not 25 minutes after).
>=20
> We will be projecting from what is on the meeting=20
> presentations site (no
> exceptions) and therefore if your slides are not there, you=20
> will be presenting to a blank screen.
>=20
> Remember that as well as people trying to address your issues=20
> in the meeting, we will have people trying to follow the=20
> discussion remotely.
> Access to the slides in advance helps everybody.
>=20
> And remember, we are looking for issues for the meeting to=20
> discuss, not tutorials on the document contents - everyone is=20
> expected to have read those.
>=20
> Regards
>=20
> Keith
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Sun Jul 22 10:19:22 2007
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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and Content Negotiation
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To rephrase what Eric is saying here in base principles: _inferring_=20
what can be done with INFO by examining supported content types is a "Do=20
What I Mean" mechanism (as opposed to a "Do What I Say" mechanism). "Do=20
What I Mean" is a well-known recipe for interoperability failure. (For a=20
well constructed discussion of this, see Jonathan's service indication=20
draft).

This is, for example, why we have an "Event" header field in RFC 3265=20
instead of simply inferring the event package from MIME type. (In fact,=20
this is why I started the draft that became 3265 in the first place --=20
with the dual uses of PINT and presence threatening to take the=20
SUBSCRIBE method down the path that INFO unfortunately ended up taking,=20
an intervention was necessary).

To re-cast Eric's example from a slightly different angle: imagine that=20
a desired usage of INFO most naturally used application/sdp as its=20
payload. How would one indicate support for such a usage?

/a

Christer Holmberg (JO/LMF) wrote:
> Hi,
>
> I still don't get it, but let's take it face-to-face at some point :)
>
> Regards,
>
> Christer=20
>
>  =20
>> -----Original Message-----
>> From: Eric Burger [mailto:eburger@bea.com]=20
>> Sent: 22. hein=E4kuuta 2007 10:14
>> To: Christer Holmberg (JO/LMF); IETF SIP
>> Subject: Re: [Sip] INFO and Content Negotiation
>>
>> That is the point - people have been saying, "I can negotiate=20
>> INFO by using Content-Type."  This is a tiny example where=20
>> using Content-Type to negotiate "INFO Packages" falls apart.
>>
>>
>> On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
>> <christer.holmberg@ericsson.com> wrote:
>>
>>    =20
>>> Why would I be expecting KPML, if I have only indicated=20
>>>      =20
>> that I support=20
>>    =20
>>> DTMF-over-INFO?
>>>
>>> If I was to expect KPML, I would first need to establish a=20
>>> subscription for KPML (or, in our "controlled network", the=20
>>> subscription would be established by default, and I would=20
>>>      =20
>> know you are going to send KPML).
>>    =20
>>> Regards,
>>>
>>> Christer
>>>
>>>      =20



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From sip-bounces@ietf.org Sun Jul 22 10:42:44 2007
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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] Comment on draft-ietf-sip-fork-loop-fix-05
Date: Sun, 22 Jul 2007 09:42:31 -0500
To: Paul Kyzivat <pkyzivat@cisco.com>
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The thing you describe is not a proxy :)

Seriously, we don't put a lot of work into specifying behavior when  
things violate the specification.

I'd be willing to add a paragraph of general tone noting that the  
loop detection mechanism breaks if you remove other people's vias.
(If there were two of the things you describe in the network,  
removing each other's vias, then there is no hope of ever detecting a  
loop).

RjS

On Jul 20, 2007, at 5:20 PM, Paul Kyzivat wrote:

> Just catching up :-(
>
> One thing occurred to me - this document goes out of its way to say  
> that a proxy need not do the loop check unless it forks. In general  
> that makes sense, but in one case it does not:
>
> Some proxies hide/suppress/obfuscate via headers, and then reverse  
> the process on responses. A proxy that does so breaks loop checking  
> by any of the proxies whose vias it hid. So I think a proxy that  
> does this MUST perform a loop check itself even if it does not fork.
>
> 	Thanks,
> 	Paul
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Eric Burger wrote:
> Well, we are only sort-of munging MIME.  The default is always RENDER.
> However, *what* is a render is an exercise for the protocol to specify.
> That is the wiggle room for SIP to say "render means by-context."
> 
> What we do not have the wiggle room for is to say the default
> content-disposition is something other than RENDER.

Well, we have *already* said that the default Content-Disposition is 
something other than "render" for certain Content-Types. E.g. for C-T of 
application/sdp the default disposition is "session". There are other 
special cases for some other content types as well.

Is that really kosher for MIME, or did we already do something bad?

In any case, what I was suggesting is that we spend some effort defining 
what "render" means for SIP.

	Paul

> On 7/22/07 12:17 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
> 
>>
>>
>> Eric Burger wrote:
>>> I would offer this is where we can "add value" to MIME for SIP usage.  Note
>>> that I do not say, "deviate from MIME."  I would suggest we NOT try to
>>> deviate from MIME in any respect.
>> Oh MIME sage, your comment is inscrutable! :-)
>>
>> I don't understand in this case where the line is between "adding value"
>> and "deviating".
>>
>> If we try to ascribe some common meaning to "render" then I believe sip
>> is already violating it. If we have thus already "deviated" from MIME,
>> which takes precedence: correcting the deviation, or backward compatibility.
>>
>> My assumption is that backward compatibility is more important.
>>
>>         Paul
>>
>>> On 7/19/07 9:59 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:
>>> [snip]
>>>> I think we just have to grandfather the use of 'refer' in SIP, and
>>>> document that the usage in sip does not mean "render", but rather means
>>>> that the usage must be determined by other context - specifically the
>>>> type of message or response and the content type. IOW, 'render' in sip
>>>> can be thought of as a peculiar spelling for 'by-context'. It might be
>>>> helpful to give guidance for development of future specs about when
>>>> 'render' should be used vs defining a new disposition.
>>> [snip]
>>>
>>>
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>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
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> 
> 
> 
> Notice:  This email message, together with any attachments, may contain information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entities,  that may be confidential,  proprietary,  copyrighted  and/or legally privileged, and is intended solely for the use of the individual or entity named in this message. If you are not the intended recipient, and have received this message in error, please immediately return this by email and then delete it.
> 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Robert Sparks wrote:
> The thing you describe is not a proxy :)
> 
> Seriously, we don't put a lot of work into specifying behavior when 
> things violate the specification.
> 
> I'd be willing to add a paragraph of general tone noting that the loop 
> detection mechanism breaks if you remove other people's vias.

That works for me.

The technique is in widespread use. Often the things doing it are 
clearly B2BUAs (though that doesn't mitigate the looping problem), but I 
have certainly heard of things that are otherwise proxies that do this. 
I think it behooves us to point out when techniques that people think 
are reasonable actually cause real problems.

> (If there were two of the things you describe in the network, removing 
> each other's vias, then there is no hope of ever detecting a loop).

Right. :-(

	Paul

> RjS
> 
> On Jul 20, 2007, at 5:20 PM, Paul Kyzivat wrote:
> 
>> Just catching up :-(
>>
>> One thing occurred to me - this document goes out of its way to say 
>> that a proxy need not do the loop check unless it forks. In general 
>> that makes sense, but in one case it does not:
>>
>> Some proxies hide/suppress/obfuscate via headers, and then reverse the 
>> process on responses. A proxy that does so breaks loop checking by any 
>> of the proxies whose vias it hid. So I think a proxy that does this 
>> MUST perform a loop check itself even if it does not fork.
>>
>>     Thanks,
>>     Paul
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 


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From sip-bounces@ietf.org Sun Jul 22 12:24:49 2007
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Some late WGLC comments on this. Don't think there are any showstoppers
here.

1)	I am not seeing the separation of UAC, UAS, and proxy
requirements required by RFC 4485. Please do not write one requirement
covering two of these entities, and ideally separate these requirements
into separate sections. To me this also means separating UAC and
registrar requirements (which for example impacts 4.1.1). The point of
this is to make it easy for an implementor of a particular item of
equipment (e.g. a terminal, or a proxy including a registrar) to clearly
identify their own requirements they have to meet.

2)	Section 3.1.3:

   In fact, implementations
   SHOULD try to establish a TLS connection when using a SIP URI.

Not sure the RFC 2119 language is appropriate here. Surely this should
be covered by more specific RFC 2119 language in section 4.

3)	Section 4.1.3:

   As a general principle, derived dialogs and transactions MUST NOT
   result in an effective downgrading of SIPS to SIP, without the
   explicit authorization of the entities involved.

Not sure how we can demonstrate true compliance against this, and
therefore do not believe the RFC 2119 language is appropriate.

4)	Section 4.1.4:

   Again, the general principle is that
   these mechanism SHOULD NOT result in an effective downgrading of SIPS
   to SIP, without the proper authorization.

Again, not sure how we can demonstrate true compliance against this, and
therefore do not believe the RFC 2119 language is appropriate.

5)	Section 4.2:

   Some proxies MAY have policies that prohibits
   sending any request over anything but TLS.

In the context, this is just explaining the previous SHOULD. I don't
think the RFC 2119 language is appropriate.

Similarly for this text in the same section:

   Some
   proxies MAY have a policy of not forwarding at all requests using a
   non-SIPS Request-URI if the UAS had registered using a SIPS Contact
   header fields.

6)	Section 4.2:

   The "transport=3Dtls" parameter must not be used by proxies that
   conform to this specification.

Is this a restatement of the text in RFC 3261 (in which case I would
prefer quoting the RFC 3261 text) and saying that there is no change
from this:

      Note that in the SIPS URI scheme, transport is independent of TLS,
      and thus "sips:alice@atlanta.com;transport=3Dtcp" and
      "sips:alice@atlanta.com;transport=3Dsctp" are both valid (although
      note that UDP is not a valid transport for SIPS).  The use of
      "transport=3Dtls" has consequently been deprecated, partly because
      it was specific to a single hop of the request.  This is a change
      since RFC 2543.

Or is it a new normative requirement in its own right, in which case the
we need "MUST NOT".

7)	Section 5.1:

   This response differs from 416 because the UAS or
   proxy recognizes the SIPS URI but it is not allowed, and because the
   UAC SHOULD NOT retry the request automatically using a SIP URI.

Is this attempting to state (in which case it should be elsewhere):

   If a 418 (SIPS Not Allowed) is received, the
   UAC SHOULD NOT retry the request automatically using a SIP URI.

Or is it just defining the semantics of the 418 in which case the RFC
2119 language is not appropriate.

8)	Section 5.2:

   The UAC SHOULD retry the request automatically using a
   SIPS URI.

Again not convinced that the RFC 2119 language is appropriate here. This
should be defining the semantics of the response code.

9)	Section 7:

   For mutual TLS,
   it means that one should be very careful that the architecture of the
   network is such that connections are made between entities that have
   access to each other's certificates.  Record-Route [RFC3261] and Path
   [RFC3327] are very useful in ensuring that previously established TLS
   connections can be re-used.  Other mechanisms may also be used in
   certain circumstances: for example, using root certificates that are
   widely recognized may allow for more easily created TLS connections.

Does this not get covered by the connect-reuse draft, if we ever get
that finished.

NITS:
-----

A)	There is significant usage of lower case RFC 2119 language,
particularly "must". I believe it is desirable to avoid this type of
usage if possible, as it only takes a slip of the editor for it to
become a "MUST" and we have a new requirement that was not agreed by the
WG. I think it is possible to find alternatives.

B)	There are a number of instances of "it" where I would prefer
"the proxy" or whatever, e.g. in structures such as:=20

   When a proxy inserts a Record-Route entry, it must take care in using
   the proper scheme=20

Should become "the proxy". I don't want people fighting their way back
in the sentence to find the real subject.

And

   If the proxy needs to reject the request for
   that reason, it MUST reject it with a 418 (SIPS Not Allowed).

Should become:

   If the proxy needs to reject the request for
   that reason, the proxy MUST reject it with a 418 (SIPS Not Allowed).

C)	Ensure that all request names, e.g "REGISTER" are followed by
"request", i.e. "REGISTER request".

D)	Section 4.1.2:

   Because registering with a SIPS Contact header field implies a
   binding of both a SIPS Contact and a corresponding SIP Contact to the
   AOR, a UA MUST NOT include both the SIPS and the SIP version of the
   same Contact header field in a REGISTER: it MUST only use the SIPS
   version in this case.

":" --> ";", and also as in B) above "it" --> "the UA"

E)	Capitalisation. "proxy", "redirect server" etc are not proper
nouns, therefore please use lower case initial letters.

F)	Section 4.1.2:

   If all the Contact header fields in a REGISTER are SIPS, a SIPS AOR
   MUST be used by the UAC in the REGISTER. If at least one of the
   Contact header fields is SIP or is neither SIP nor SIPS (e.g.,
   mailto, tel, http, https), a SIP AOR MUST also be used by the UAC.=20

Rewrite as

   If all the Contact header fields in a REGISTER request are SIPS, a
the UAC=20
   MUST use a SIPS AOR in the REGISTER request. If at least one of the
   Contact header fields is SIP or is neither SIP nor SIPS (e.g.,
   mailto, tel, http, https), the UAC MUST also use a SIP AOR.

G)	Section 4.1.3:

   If the top Route header field
   does not contain a SIPS URI, the Contact header field MUST be a SIP
   URI, even if the request is sent over a secure transport (e.g., the
   first hop could be re-using a TLS connection to the proxy as would be
   the case with [I-D.ietf-sip-outbound]).

Rewrite to clearly indicate a UAC requirement, ie. "the UAC must..."

H)	Section 4.1.4:

   If that initial dialog was
   established using SIPS, then the new one MUST NOT be established
   using SIP, unless there is an explicit authorization given by the
   recipient of the REFER.

Rewrite to clearly indicate a UAC requirement, ie. "the UAC must..."

I)	Appendix B:

      If the address-of-record in the To header field of a REGISTER
      request is a SIPS URI, then any Contact header field value in the
      request MUST also be SIPS URIs.

Rewrite as UAC requirement:

      If the address-of-record in the To header field of a REGISTER
      request is a SIPS URI, then the UAC MUST also include only SIPS
URIs in any Contact header field value in the
      request.


Regards

Keith


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Dale.Worley@comcast.net wrote:
>    From: Volker Hilt <volkerh@bell-labs.com>
> 
>    I've submitted a new version of the "Essential Correction to the Session 
>    Initiation Protocol (SIP) 503 (Service Unavailable) Response":
>    http://www.ietf.org/internet-drafts/draft-hilt-sip-correction-503-01.txt
> 
> This work is important, but I don't see how it fits in the category
> "essential correction", as a SIP system can function without it.
> Perhaps you could add a section explaining why it is an essential
> correction.
> 
The problem is that, with the current mechanism, a SIP system won't 
function properly when it is in overload in many deployments. Since it 
seems close to impossible to dimension a SIP system so that it will 
never experience overload, I think fixing this is important for the 
functioning of a SIP system.

I'll add a section explaining this to the next rev of the draft.

Thanks,

Volker



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



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At 11:15 PM 7/21/2007, DRAGE, Keith \(Keith\) wrote:
>1)      I am not convinced this document does updated RFC 4412.

this ID is creating the OPTIONAL delimiter, which would be opaque to 
implementers otherwise, and potentially causing implementations to 
not process the header (therefore message) correctly.

I know this is a *really* small update, but since this delimiter is 
splitting the namespace into two-parts, when the '-' character 
appears, it requires coders to code differently according to this 
document - were they to chose to support this extension to RFC4412. 
Coders will look for the '-' character anywhere in a namespace, and 
have special process(es) for namespaces that have this delimiter.

>RFC 4412
>creates the namespace. This document merely adds new items to the IANA
>registry created by that document.

and the delimiter

>The document header should be
>modified accordingly, and the following text in the abstract also
>corrected:
>
>    This
>    document intends to update RFC 4412, as a Proposed Standard document
>    if published by the RFC-Editor.

once we agree, yes


>2)      I believe the abstract could usefully summarise the new
>namespaces that are created (in the same way as most drafts that define
>new headers identify them in the abstract).

I'm open to how to summarize the new namespaces. Should the abstract 
state the namespaces are for DISA?


>3)      In section 3, I do not see the need to include the following
>text in the document, as it is already in RFC 4412, for which all the
>requirements apply, not just this one.
>
>    As stated in Section 9 of RFC 4412 [RFC4412], an IANA registered
>    namespace SHOULD NOT change the number, and MUST NOT change the
>    relative priority order, of its assigned priority-values.

fair point


>4)      NIT. Tag all mentions of RFC4412 as references.

will do



>Regards
>
>Keith


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From sip-bounces@ietf.org Sun Jul 22 20:27:23 2007
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Subject: RE: [Sip] INFO and Content Negotiation
Date: Mon, 23 Jul 2007 02:27:11 +0200
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Adam Roach" <adam@nostrum.com>
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Hi,

I think it is clearly said that INFO is not to be used to directly =
modify the SIP session, so unless there is an application on top of the =
SIP session that understands this information it would simply be =
rejected.

Regards,

Christer


=20

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]=20
> Sent: 22. hein=E4kuuta 2007 17:19
> To: Christer Holmberg (JO/LMF)
> Cc: Eric Burger; IETF SIP
> Subject: Re: [Sip] INFO and Content Negotiation
>=20
> To rephrase what Eric is saying here in base principles:=20
> _inferring_ what can be done with INFO by examining supported=20
> content types is a "Do What I Mean" mechanism (as opposed to=20
> a "Do What I Say" mechanism). "Do What I Mean" is a=20
> well-known recipe for interoperability failure. (For a well=20
> constructed discussion of this, see Jonathan's service=20
> indication draft).
>=20
> This is, for example, why we have an "Event" header field in=20
> RFC 3265 instead of simply inferring the event package from=20
> MIME type. (In fact, this is why I started the draft that=20
> became 3265 in the first place -- with the dual uses of PINT=20
> and presence threatening to take the SUBSCRIBE method down=20
> the path that INFO unfortunately ended up taking, an=20
> intervention was necessary).
>=20
> To re-cast Eric's example from a slightly different angle:=20
> imagine that a desired usage of INFO most naturally used=20
> application/sdp as its payload. How would one indicate=20
> support for such a usage?
>=20
> /a
>=20
> Christer Holmberg (JO/LMF) wrote:
> > Hi,
> >
> > I still don't get it, but let's take it face-to-face at=20
> some point :)
> >
> > Regards,
> >
> > Christer
> >
> >  =20
> >> -----Original Message-----
> >> From: Eric Burger [mailto:eburger@bea.com]
> >> Sent: 22. hein=E4kuuta 2007 10:14
> >> To: Christer Holmberg (JO/LMF); IETF SIP
> >> Subject: Re: [Sip] INFO and Content Negotiation
> >>
> >> That is the point - people have been saying, "I can=20
> negotiate INFO by=20
> >> using Content-Type."  This is a tiny example where using=20
> Content-Type=20
> >> to negotiate "INFO Packages" falls apart.
> >>
> >>
> >> On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
> >> <christer.holmberg@ericsson.com> wrote:
> >>
> >>    =20
> >>> Why would I be expecting KPML, if I have only indicated
> >>>      =20
> >> that I support
> >>    =20
> >>> DTMF-over-INFO?
> >>>
> >>> If I was to expect KPML, I would first need to establish a=20
> >>> subscription for KPML (or, in our "controlled network", the=20
> >>> subscription would be established by default, and I would
> >>>      =20
> >> know you are going to send KPML).
> >>    =20
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>      =20
>=20
>=20


_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Sun Jul 22 20:37:24 2007
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Subject: RE: [Sip] INFO and Content Negotiation
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	<7374777208BDC7449D5620EF9423256707479CEF@esealmw113.eemea.ericsson.se>
From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Adam Roach" <adam@nostrum.com>
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Hi,

I forgot the part about the indication of the support.

I agree that Accept: application/sdp itself may not be enough to =
indicate the support for the specific usage.

So, either you could have e.g. an option-tag, some application =
identifier, an extention to the Accept header - or we would simply say =
that INFO cannot be used to transport SDP (as I've said many times, I am =
not proposing to allow to use INFO for whatever people want to).

Regards,

Christer=20


> -----Original Message-----
> From: Christer Holmberg (JO/LMF)=20
> [mailto:christer.holmberg@ericsson.com]=20
> Sent: 23. hein=E4kuuta 2007 3:27
> To: Adam Roach
> Cc: IETF SIP
> Subject: RE: [Sip] INFO and Content Negotiation
>=20
>=20
> Hi,
>=20
> I think it is clearly said that INFO is not to be used to=20
> directly modify the SIP session, so unless there is an=20
> application on top of the SIP session that understands this=20
> information it would simply be rejected.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> =20
>=20
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@nostrum.com]
> > Sent: 22. hein=E4kuuta 2007 17:19
> > To: Christer Holmberg (JO/LMF)
> > Cc: Eric Burger; IETF SIP
> > Subject: Re: [Sip] INFO and Content Negotiation
> >=20
> > To rephrase what Eric is saying here in base principles:=20
> > _inferring_ what can be done with INFO by examining=20
> supported content=20
> > types is a "Do What I Mean" mechanism (as opposed to a "Do=20
> What I Say"=20
> > mechanism). "Do What I Mean" is a well-known recipe for=20
> > interoperability failure. (For a well constructed=20
> discussion of this,=20
> > see Jonathan's service indication draft).
> >=20
> > This is, for example, why we have an "Event" header field=20
> in RFC 3265=20
> > instead of simply inferring the event package from MIME type. (In=20
> > fact, this is why I started the draft that became 3265 in the first=20
> > place -- with the dual uses of PINT and presence=20
> threatening to take=20
> > the SUBSCRIBE method down the path that INFO unfortunately ended up=20
> > taking, an intervention was necessary).
> >=20
> > To re-cast Eric's example from a slightly different angle:=20
> > imagine that a desired usage of INFO most naturally used=20
> > application/sdp as its payload. How would one indicate support for=20
> > such a usage?
> >=20
> > /a
> >=20
> > Christer Holmberg (JO/LMF) wrote:
> > > Hi,
> > >
> > > I still don't get it, but let's take it face-to-face at
> > some point :)
> > >
> > > Regards,
> > >
> > > Christer
> > >
> > >  =20
> > >> -----Original Message-----
> > >> From: Eric Burger [mailto:eburger@bea.com]
> > >> Sent: 22. hein=E4kuuta 2007 10:14
> > >> To: Christer Holmberg (JO/LMF); IETF SIP
> > >> Subject: Re: [Sip] INFO and Content Negotiation
> > >>
> > >> That is the point - people have been saying, "I can
> > negotiate INFO by
> > >> using Content-Type."  This is a tiny example where using
> > Content-Type
> > >> to negotiate "INFO Packages" falls apart.
> > >>
> > >>
> > >> On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
> > >> <christer.holmberg@ericsson.com> wrote:
> > >>
> > >>    =20
> > >>> Why would I be expecting KPML, if I have only indicated
> > >>>      =20
> > >> that I support
> > >>    =20
> > >>> DTMF-over-INFO?
> > >>>
> > >>> If I was to expect KPML, I would first need to establish a=20
> > >>> subscription for KPML (or, in our "controlled network", the=20
> > >>> subscription would be established by default, and I would
> > >>>      =20
> > >> know you are going to send KPML).
> > >>    =20
> > >>> Regards,
> > >>>
> > >>> Christer
> > >>>
> > >>>      =20
> >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] INFO and Content Negotiation
Date: Mon, 23 Jul 2007 02:40:25 +0200
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "Adam Roach" <adam@nostrum.com>
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=20
...and I guess you would also use another content-disposition value than =
"session" in your usage (again, since INFO cannot be used to directly =
modify the SIP session).

NOW I'm done with the reply to your e-mail ;)

Regards,

Christer

> -----Original Message-----
> From: Christer Holmberg (JO/LMF)=20
> [mailto:christer.holmberg@ericsson.com]=20
> Sent: 23. hein=E4kuuta 2007 3:37
> To: Adam Roach
> Cc: IETF SIP
> Subject: RE: [Sip] INFO and Content Negotiation
>=20
>=20
> Hi,
>=20
> I forgot the part about the indication of the support.
>=20
> I agree that Accept: application/sdp itself may not be enough=20
> to indicate the support for the specific usage.
>=20
> So, either you could have e.g. an option-tag, some=20
> application identifier, an extention to the Accept header -=20
> or we would simply say that INFO cannot be used to transport=20
> SDP (as I've said many times, I am not proposing to allow to=20
> use INFO for whatever people want to).
>=20
> Regards,
>=20
> Christer=20
>=20
>=20
> > -----Original Message-----
> > From: Christer Holmberg (JO/LMF)
> > [mailto:christer.holmberg@ericsson.com]
> > Sent: 23. hein=E4kuuta 2007 3:27
> > To: Adam Roach
> > Cc: IETF SIP
> > Subject: RE: [Sip] INFO and Content Negotiation
> >=20
> >=20
> > Hi,
> >=20
> > I think it is clearly said that INFO is not to be used to directly=20
> > modify the SIP session, so unless there is an application on top of=20
> > the SIP session that understands this information it would=20
> simply be=20
> > rejected.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> > =20
> >=20
> > > -----Original Message-----
> > > From: Adam Roach [mailto:adam@nostrum.com]
> > > Sent: 22. hein=E4kuuta 2007 17:19
> > > To: Christer Holmberg (JO/LMF)
> > > Cc: Eric Burger; IETF SIP
> > > Subject: Re: [Sip] INFO and Content Negotiation
> > >=20
> > > To rephrase what Eric is saying here in base principles:=20
> > > _inferring_ what can be done with INFO by examining
> > supported content
> > > types is a "Do What I Mean" mechanism (as opposed to a "Do
> > What I Say"=20
> > > mechanism). "Do What I Mean" is a well-known recipe for=20
> > > interoperability failure. (For a well constructed
> > discussion of this,
> > > see Jonathan's service indication draft).
> > >=20
> > > This is, for example, why we have an "Event" header field
> > in RFC 3265
> > > instead of simply inferring the event package from MIME type. (In=20
> > > fact, this is why I started the draft that became 3265 in=20
> the first=20
> > > place -- with the dual uses of PINT and presence
> > threatening to take
> > > the SUBSCRIBE method down the path that INFO=20
> unfortunately ended up=20
> > > taking, an intervention was necessary).
> > >=20
> > > To re-cast Eric's example from a slightly different angle:=20
> > > imagine that a desired usage of INFO most naturally used=20
> > > application/sdp as its payload. How would one indicate=20
> support for=20
> > > such a usage?
> > >=20
> > > /a
> > >=20
> > > Christer Holmberg (JO/LMF) wrote:
> > > > Hi,
> > > >
> > > > I still don't get it, but let's take it face-to-face at
> > > some point :)
> > > >
> > > > Regards,
> > > >
> > > > Christer
> > > >
> > > >  =20
> > > >> -----Original Message-----
> > > >> From: Eric Burger [mailto:eburger@bea.com]
> > > >> Sent: 22. hein=E4kuuta 2007 10:14
> > > >> To: Christer Holmberg (JO/LMF); IETF SIP
> > > >> Subject: Re: [Sip] INFO and Content Negotiation
> > > >>
> > > >> That is the point - people have been saying, "I can
> > > negotiate INFO by
> > > >> using Content-Type."  This is a tiny example where using
> > > Content-Type
> > > >> to negotiate "INFO Packages" falls apart.
> > > >>
> > > >>
> > > >> On 7/22/07 3:10 AM, "Christer Holmberg (JO/LMF)"
> > > >> <christer.holmberg@ericsson.com> wrote:
> > > >>
> > > >>    =20
> > > >>> Why would I be expecting KPML, if I have only indicated
> > > >>>      =20
> > > >> that I support
> > > >>    =20
> > > >>> DTMF-over-INFO?
> > > >>>
> > > >>> If I was to expect KPML, I would first need to establish a=20
> > > >>> subscription for KPML (or, in our "controlled network", the=20
> > > >>> subscription would be established by default, and I would
> > > >>>      =20
> > > >> know you are going to send KPML).
> > > >>    =20
> > > >>> Regards,
> > > >>>
> > > >>> Christer
> > > >>>
> > > >>>      =20
> > >=20
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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



From sip-bounces@ietf.org Mon Jul 23 07:03:46 2007
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From: sunqian 32328 <sunqian@huawei.com>
Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
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Hi,

Could different SIP methods (INVITE, MESSAGE) be given different Levels of Support for Multipart?

Best Regards,
Qian Sun

----- Original Message -----
From: Paul Kyzivat <pkyzivat@cisco.com>
Date: Sunday, July 22, 2007 12:21 pm
Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01

> One of the tasks of the draft is to make support for at least some 
> subtypes of multipart (e.g. mixed) mandatory.
> 
> I hope this is not a suggestion to make dereferencing of external 
> bodies 
> mandatory.
> 
> If the request is just to include some clarification of the use 
> and 
> handling of external bodies in this draft then I guess that does 
> sound 
> useful.
> 
> 	Paul
> 
> Eric Burger wrote:
> > Seems like a good idea.  However, I would not go so far as to 
> mandate> iterative evaluation.  I would offer that the UA 
> evaluates as needed.
> > 
> > 
> > On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:
> > 
> >> Hi,
> >> I suggest Content Indirection Mechanism defined in rfc4483 
> should also be
> >> considered in this draft.
> >>
> >> It is a little special, an example from rfc4483:
> >> 6.1.  Single Content Indirection
> >>
> >>            INVITE sip:boromir@example.com SIP/2.0
> >>            Accept: message/external-body application/sdp
> >>            Content-Type: message/external-body;
> >>                 ACCESS-TYPE=URL;
> >>                 
> URL="http://www.example.net/party/06/2002/announcement";>>         
>        EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
> >>                 size=231
> >>            Content-Length: 105
> >>
> >>            Content-Type: application/sdp
> >>            Content-Disposition: session
> >>            Content-ID: <4e5562cd1214427d@example.net>
> >>
> >> In addition, what happens to the INVITE request if the 
> recipient can not get
> >> the indirect content?
> >>
> >> Nits: a letter "l" is missing in the title.
> >> Internet-Draft         Message Body Hand"l"ing in SIP           
>   May 2007
> >>
> >> Cheers,
> >> Qian
> >>
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Friday, July 20, 2007 2:30 AM
> >> To: Gonzalo Camarillo
> >> Cc: IETF SIP List
> >> Subject: Re: [Sip] comments on draft-camarillo-sip-body-
> handling-01
> >>
> >> Just one comment in addition to what Gonzalo said:
> >>
> >> Gonzalo Camarillo wrote:
> >>
> >>>> The use case described for multipart/alternative - new SDP 
> mechanisms>>>> - seems kind of bogus. We've decided not to pursue 
> that. Do we have a
> >>>> REAL use case for this?
> >>> AFAICT, we have decided not to pursue SDPng but we still need a
> >>> mechanism in order to be able to, at some point, migrate to 
> new session
> >>> description formats. The current consensus, as I understand 
> it, is that
> >>> multipart/alternative should not be used to provide 
> alternative SDP
> >>> descriptions, but that it could be used to provide alternative 
> session>>> descriptions written in different formats.
> >> The other classic example, because it is just like email, is 
> MESSAGE>> with a multipart/alternative containing plain text and html.
> >>
> >> Paul
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > Notice:  This email message, together with any attachments, may 
> contain information  of  BEA Systems,  Inc.,  its subsidiaries  
> and  affiliated entities,  that may be confidential,  proprietary, 
> copyrighted  and/or legally privileged, and is intended solely 
> for the use of the individual or entity named in this message. If 
> you are not the intended recipient, and have received this message 
> in error, please immediately return this by email and then delete it.
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 


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Subject: [Sip] comments on draft-ietf-sip-sips
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My two main large comments are organizational, but they are important:

1. the draft reads like a union of an informational overview of sips and 
a normative specification of sips. If this draft is meant to normatively 
describe sips, section 3 needs to turn into a proper overview of 
operations section. Right now it is enormous, way out of wack for what 
it should be.

2. the normative rules are very hard to follow. They need to be split up 
in the way described in rfc4596, and have clear rules for UAC, UAS and 
proxy. They should separately handle dialog forming requests, mid-dialog 
requests, registrations, and non-dialog requests for all three.

Some more specific comments:


>  While SIPS is useful to request that a resource be contacted
>    securely, it is not useful as an indication that a resource was in
>    fact contacted securely.

This is not quite true; its an indication that, if you trust all
proxies, the resource will be contacted securely.

> There is
>    currently no mechanism to provide an indication of end-to-end
>    security for SIP. 

I think its worth noting here that this is a consequence of the
difficulty in doing this. The fact that we have both signaling and
media, that the connectivity is between two endpoints and not just
client/server, and things like conferences and PSTN gateways make this
incredibly complicated - far more than a web server.

> Some have asked why is SIPS useful in a global open environment such
>    as the Internet, if (when used in a Request-URI) it is not an
>    absolute guarantee that the request will in fact be delivered over
>    TLS on each hop?  Why is SIPS any different than just using TLS
>    transport with SIP? 

there are two separate questions here and your text that follows answers
only the second.

To answer the first, I think its important to point out that, a large
number of deployments make an assumption that the servers are trusted.
In such a deployment, sips provides protection against a large number of
attacks that would otherwise be possible. It prevents against
eavesdropping of signaling, modification of signaling, and other attacks
by any outsider. The only attack it does not mitigate is ones where the
network infrastructure itself is the attacker. However, the view is that
its level of security is satisfactory for many applications.

> It is outside of the scope of this document to specify what
>    happens when a SIPS URI identifies a UAS resource that "maps" outside
>    of the SIP network, for example, to other networks such as the PSTN.

A SIPS URI never identifies a resource outside of SIP; its definition is
that it identifies a sip resource. That resource may just happen to be a
number on a PSTN gateway, which the gateway then chooses to contact
through the PSTN. The main point is that SIPS protects only within the
SIP network, from UAC to UAS.


Section 3.2.1 is really short and its not clear why its not just folded
into 3.2, or else 3.2 should be split up more.

> Since upgrading from SIP to SIPS is allowed in other circumstances
>    (e.g., a user "guessing" a SIPS AOR from a SIP AOR on a business
>    card), 

I think its important to be clear here that this upgrading is by a human
user and not by a proxy.

Also the term guessing is wrong since its not a guess at all that this
is the right URI. It is. Its only a guess about whether it will work. Of
course you can't know in advance if a call to a non-SIPS URI will work
either. So I'd suggest nuking the 'guessing' term.

Section 3.3 is pretty clear on downgrading but is very vague on
upgrading. Some clear statements here are needed.

> to describe how to achieve backward compatibility.  If a proxy
>    conforms to this specification, the procedures in this section are
>    not used.

If we have eliminated changing from SIP to SIPS we should nuke this
section. Anyway double record-routing is now documented in the
record-route-fix draft so why do we need it here.

> The reinstatement of the transport=tls parameter, or of an
>    alternative mechanism for indicating the use of the TLS on a single
>    hop in a URI, is outside the scope of this specification (see
>    Appendix A.3).

Its really odd that we have a whole section explaining how it works,
followed by a statement that says ignore the previous section since we
are not re-instating it. If its in, we should describe it. If its out,
we should say so, and thats it. I personally think real world experience
has shown it useful and we should keep it.


> When presented with a SIPS URI, a UAC or UAS MUST NOT change it to a
>    SIP URI.  For example, if a directory entry includes a SIPS AOR, the
>    UAC must not send requests to that AOR using a SIP Request-URI.
>    Similarly, if a user reads a business card with a SIPS URI, he should
>    not infer a SIP URI.  If a 3XX response includes a SIPS Contact
>    header field, the UAC MUST NOT replace it with a SIP Request-URI
>    (e.g., by replacing the SIPS scheme with a SIP scheme) when sending a
>    request as a result of the redirection.


I think with user interaction this is fine, and certainly cannot be 
prevented.

> If a UAS does not wish to be reached with a SIPS URI but only with a
>    SIP URI, it SHOULD respond with a 418 (SIPS Not Allowed). 

Respond to what? I think you need to say, "request whose request-URI is 
a SIP URI"

> Upon receiving a 416 or 418, a UAC MUST NOT re-
>    attempt the request by automatically replacing the SIPS scheme with a
>    SIP scheme as described in [RFC3261]/8.1.3.5 as it would be a
>    security vulnerability.  If the UAC does re-attempt the call with a
>    SIP URI, it SHOULD get a confirmation from the user to authorize re-
>    initiating the session with a SIP Request-URI instead of a SIPS
>    Request-URI.

These two sentences are contradictory.


Section 4 really needs to have separate behaviors for UAC and UAS, proxy 
and registrar.

> The UAC registers Contacts header fields to either a SIPS or a SIP
>    AOR.  From a routing perspective, it does not matter which one is
>    used for registration as they identify the same resource.


the part, "which one is used" could refer to either contact or AOR. You 
should clarify you mean AOR in the To header field.

> Similarly, a UA SHOULD NOT register both a SIP
>    Contact header field and a SIPS Contact header field in separate
>    registrations as the SIP Contact header field would be superfluous.

What should proxy routing logic be in this case? Should clarify.

> If all the Contact header fields in a REGISTER are SIPS, a SIPS AOR
>    MUST be used by the UAC in the REGISTER.  If at least one of the
>    Contact header fields is SIP or is neither SIP nor SIPS (e.g.,
>    mailto, tel, http, https), a SIP AOR MUST also be used by the UAC.
>    However, the registrar MUST treat the SIP and SIPS schemes of the AOR
>    the same way (i.e., it MUST not care if it is SIP or SIPS).  These
>    are purely mechanical rules with no influence on routing.

The MUST requirement for setting the AOR in the To header field doesn't 
make sense when combined with the requirement that proxies MUST treat 
sip and sips the same. The requirement on the AOR in the To from a UAC 
probably should be a SHOULD. Also you should be clear that when you say 
AOR you mean To header field since there is an AOR in the From header 
field also.

> Furthermore, it is a matter of local policy for a UA to accept
>    incoming requests addressed to a URI scheme that does not correspond
>    to what it used for registration.  For example, a UA with a policy of
>    "always SIPS" would address the Registrar using a SIPS Request-URI
>    over TLS, 

This is the first mention of setting the R-URI in REGISTER. I think its 
important. Indeed this paragraph provides the clearest guidance on what 
to do, and its not normative text..

> A registrar MUST only accept a binding to a SIPS Contact header field
>    if all the appropriate URIs are of the SIPS scheme, otherwise there
>    could be an inadvertent binding of a secure resource (SIPS) to an
>    unsecured one (SIP).  This includes the Request-URI, the Contacts and
>    all the Path headers, but does not include the From and To headers.
>    If the URIs are not of the proper SIPS scheme, the registrar MUST
>    reject the REGISTER with a 419 (SIPS Required).

You need to define 'appropriate'. Please state explicit cases and 
behaviors that are required.

> If the Request-URI in a request that initiates a dialog is a SIP URI,
>    then the UAC needs to be careful about what to use in the Contact
>    header field (in case Record-Route is not used for this hop).  If the
> 
> 
> 
> Audet                   Expires December 27, 2007              [Page 17]
> 
> Internet-Draft                    SIPS                         June 2007
> 
> 
>    Contact header field was a SIPS URI, it would mean that it would only
>    accept mid-dialog requests that are sent over secure transport on
>    each hop. 

I thought we were going to forbid this case. If the R-URI is SIP, then 
Contact, Route, etc. are all SIP. If the R-URI is SIPS, then Route and 
Contact are SIPS. Mixed modes not allowed.

> When a target refresh occurs within a dialog (e.g., re-INVITE,
>    UPDATE), the UAC and UAS MUST include a Contact header field with a
>    SIPS URI if the original request used a SIPS Request-URI.

You'll get this for free as a side effect of normal dialog processing rules.

> Note that a REFER may also be used for referring to resources that do
>    not result in dialogs being created.  In fact, a REFER may be used to
>    point to resources that are of a different type than the original one
>    (i.e., not SIP or SIPS).  Please see [RFC3515]/5.2 for security
>    considerations related to this.

So what is the normative requirement here. Is it allowed to send a REFER 
on a SIPS dialog, to an http URI?

> If the wrong scheme is received in the GRUU (which would be an error
>    in the registrar), the UAC SHOULD treat it as if the proper scheme
>    was used (i.e., it SHOULD replace the scheme with the proper scheme
>    before using the GRUU).

what is the value in having this behavior? I think handling of errors in 
registrar like this is out of scope.

> When a proxy receives a request with a SIP Request-URI, the proxy
>    MUST NOT forward the request to a SIPS Request-URI.  If the target
>    UAS had registered previously using a SIPS Contact header field, and
>    the proxy decides to forward the request, it MUST replace that SIPS
>    scheme with a SIP scheme while leaving the rest of the URI as is, and
>    use the resulting URI as the Request-URI of the forwarded request.
>    The proxy MUST use TLS to forward the request to the UAS.  

when the R-URI is SIP, you MUST use TLS??

> 4.2.  Proxy Behavior

This section is missing an essential behavior somewhere, that needs to 
be very clear that when targeting a SIPS URI, you MUST send over TLS.

> Using a Redirect Server with TLS instead of using a Proxy has some
>    limitations that have to be taken into account.  Since there no pre-
>    established connection between the Proxy and the UAS (such as with
>    [I-D.ietf-sip-outbound]), it is only appropriate for scenarios where
>    inbound connections are allowed.

I don't follow this. You can have redirects and redirect servers with 
pre-established connections. What use case are you referring to here?
 From the paragraph below it seems you are assuming that the redirect is 
at the home proxy, redirecting to the registered UA's Contact. I agree 
this is not likely to work but its not likely to work without sips either.

> When a redirect server receives a request with a SIP Request-URI, the
>    redirect server MAY redirect with a 3XX response to either a SIP or a
>    SIPS Contact header field.  If the target UAS had registered
>    previously using a SIPS Contact header field, the redirect server
>    SHOULD return a SIPS Contact header field if it is in an environment
>    where TLS is usable (as described in the previous paragraph).  If the
>    target UAS had registered previously using a SIP Contact header
>    field, the redirect server MUST return a SIP Contact header field in
>    a 3XX response if it redirects the request.
> 
>    When a redirect server receives a request with a SIPS Request-URI,
>    the redirect server MAY redirect with a 3XX response to either a SIP
>    or a SIPS Contact header field.  If the target UAS had registered
>    previously using a SIPS Contact header field, the redirect server
>    SHOULD return a SIPS Contact header field if it is in an environment
>    where TLS is usable.  If the target UAS had registered previously
>    using a SIP Contact header field, the redirect server MUST return a
>    SIP Contact header field in a 3XX response if it chooses to redirect;
>    otherwise it may reject the request with a 418 (SIPS Not Allowed).

these two paragraphs are nearly identical. Should merge them or something.

Also this section treats only the case of a redirect server that is 
redirecting to registered contacts. This never really happens. most 
redirect servers are things like route servers that return PSTN gateway 
routes. You should handle this case and I think say that a server 
receiving a SIPS r-uri SHOULD redirect to SIPS.

> The UAS or proxy cannot process the request because the SIPS scheme
>    is not allowed.  This response differs from 416 because the UAS or
>    proxy recognizes the SIPS URI but it is not allowed, and because the
>    UAC SHOULD NOT retry the request automatically using a SIP URI.

why shouldn't it retry?



* draft should probably talk about the tel URI, and discuss how to handle it




Minor/Nits:

> In this model, both the TLS client and the TLS server provide a
>    certificate in the TLS handshake phase.  When no TLS connection is in
>    place, the UAC would therefore take on the role of the TLS Client and
>    the UAS would therefore take on the role of the TLS server.  Since in
>    SIP, a UA or a Proxy act both as UAC and UAS depending on if they are
>    sending or receiving requests, the symmetrical nature of mutual TLS
>    is very convenient. 

This paragraph is confusing. The discussion on UAC and UAS is misleading
since its only true for UA to UA communications. Also a proxy is not a
UAC or UAS; the terms are client and server.

>    UA a                        Proxy                         UA b
>
>          -------REQUEST sips:-------->-------REQUEST sip:--------->
>                                         Record-Route: <sip:p;lr>,
>                                                       <sips:p;lr>
>
>          <------Response sips:-------<-------Response sip:---------
>           Record-Route: <sip:p;lr>      Record-Route: <sip:p;lr>,
>                         <sips:p;lr>                   <sips:p;lr>
>
>                       Record routing from SIPS to SIP

The terms 'Response sips' and 'Response sip' are kind of meaningless
since there is no such URI in the response. Suggest you clarify instead
where there is TLS and relabel REQUEST to REQUEST-URI.

> When [I-D.ietf-sip-outbound] is used, a transport parameter in a
>    REGISTER message would normally be ignored since the existing
>    connection is re-used instead.

This is true for Contact, not for other fields.

> In a Contact header field in a dialog-creating request, it would only
>    be useful in the cases where Record-Route is not used.  However, as
>    described in this specification, this is extremely unlikely to work
>    because of the difficulty in establishing the TLS connection between
>    peers (i.e., they would need to have certificates).

too many 'its' and 'this' in this paragraph - needs clarification.

> The UAC registers Contacts header fields to either a SIPS or a SIP
>    AOR. 

s/contacts/contact
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org Mon Jul 23 09:57:13 2007
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Date: Mon, 23 Jul 2007 08:56:53 -0500
From: Peili Xu <xupeili@huawei.com>
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Hi Jonathan et al,

After reviewing the draft and mailing list discussion, I'd like to share my
view for your consideration.

Assuming we could group the problem statements into 3 categories: 
a) Conversion for contact binding; b) conversion for routing resolution and
c) conversion for services.

1: I'd suggest using P-Called-Party-ID to for category a) problem and loose
routing for the other 2.

2: When would proxy apply loose route after conversion for category 2/3 is
subject to local policy or Some kind of indication provided by external
conversion service. 

3: How to define a uri parameter for sub-address purpose is somewhat
independent from the draft to me.
But it's interesting. I'd suggest specifying it separately and with thorough
considerations for how to implement sub-address like services in SIP.

Best regards
Peili




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=20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Friday, July 20, 2007 11:31 AM
> To: Stucker, Brian (RICH1:AR00)
> Cc: Rohan Mahy; Cullen Jennings; sipping@ietf.org; Audet,=20
> Francois (SC100:3055); cboulton@ubiquitysoftware.com;=20
> Jonathan Rosenberg; Gonzalo Camarillo
> Subject: Re: Another comment on the outbound draft
>=20
>=20
> On Jul 19, 2007, at 10:47 PM, Brian Stucker wrote:
>=20
> > Rohan/Cullen,
> >
> > One other comment on the outbound draft... I think a=20
> statement needs=20
> > to be added to the draft to cover the following situation:
> >
> > Let's say a UAC is registering from behind a NAT. If the outbound=20
> > proxy does not support outbound (i.e. does not put 'ob' in the Path=20
> > header) then the resulting registration will contain an=20
> address that=20
> > is likely not reachable by the local outbound proxy after the NAT=20
> > binding has expired. The UAC can detect that this may occur=20
> by looking=20
> > for the 'outbound' tag in the supported header of the registration=20
> > response and act accordingly. However, in section 4.2 the=20
> text in the=20
> > third paragraph on page 15 shows that checking the=20
> supported header to=20
> > see if outbound processing occurred is only a MAY instead of a MUST.
> >
> > I believe that the UAC must do this checking unless they know for a=20
> > fact that their contact address is globally routable (or at least=20
> > reachable by the local outbound proxy at all times), and if not,=20
> > re-register using their server reflexive address in the contact URI=20
> > via STUN or derived through some other means (observation of=20
> > received/rport values in the registration response Via header, etc).
>=20
> Hi Brian,
>=20
> Originally I was thinking that the UAC might not care.  If=20
> the UAC already set its Contact prepared for the possibility=20
> of both outbound and non-outbound cases, it won't need to do=20
> anything differently in its initial REGISTER request. =20
> However, the UAC MUST check if the 'outbound' tag was in the=20
> Require header (see Adam's note on tags) to see if outbound=20
> was applied, so it can add or not add the 'ob' =20
> parameter to the UACs Contact URI.
>=20
> thanks,
> -rohan
>=20

Hi Rohan,

Sorry for the delay in replying.

There's a blurb in section 4.3 (third paragraph) that seems to throw a
possible wrench into that logic...=20

	The contact is formed normally in that the UAC uses the IP
address of
   the device (even if the device is behind a NAT).

The section on registrations (4.2) does not state anything in particular
about the content of the Contact URI other than the parameters that are
specific to the outbound draft itself. So, if in general, the
expectation (from section 4.3) is that the contact is composed of the IP
address of the device (wondering why it shouldn't generally be the
contact it registered) and there's nothing stated in 4.2 specifically,
it may follow that the UAC implementation doesn't consider the case I am
raising and is left without a usable registration.

Regards,
Brian


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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: sunqian 32328 <sunqian@huawei.com>
Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
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I find it quite plausible that an implementation might support multipart 
for some methods and not for others. Currently it is 
difficult/impossible to reflect that distinction in signaling. So far we 
have punted making changes to indicate such distinctions - thinking that 
doing so is not worth the trouble.

	Paul

sunqian 32328 wrote:
> Hi,
> 
> Could different SIP methods (INVITE, MESSAGE) be given different Levels of Support for Multipart?
> 
> Best Regards,
> Qian Sun
> 
> ----- Original Message -----
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Date: Sunday, July 22, 2007 12:21 pm
> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
> 
>> One of the tasks of the draft is to make support for at least some 
>> subtypes of multipart (e.g. mixed) mandatory.
>>
>> I hope this is not a suggestion to make dereferencing of external 
>> bodies 
>> mandatory.
>>
>> If the request is just to include some clarification of the use 
>> and 
>> handling of external bodies in this draft then I guess that does 
>> sound 
>> useful.
>>
>> 	Paul
>>
>> Eric Burger wrote:
>>> Seems like a good idea.  However, I would not go so far as to 
>> mandate> iterative evaluation.  I would offer that the UA 
>> evaluates as needed.
>>>
>>> On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:
>>>
>>>> Hi,
>>>> I suggest Content Indirection Mechanism defined in rfc4483 
>> should also be
>>>> considered in this draft.
>>>>
>>>> It is a little special, an example from rfc4483:
>>>> 6.1.  Single Content Indirection
>>>>
>>>>            INVITE sip:boromir@example.com SIP/2.0
>>>>            Accept: message/external-body application/sdp
>>>>            Content-Type: message/external-body;
>>>>                 ACCESS-TYPE=URL;
>>>>                 
>> URL="http://www.example.net/party/06/2002/announcement";>>         
>>        EXPIRATION="Sat, 20 Jun 2002 12:00:00 GMT";
>>>>                 size=231
>>>>            Content-Length: 105
>>>>
>>>>            Content-Type: application/sdp
>>>>            Content-Disposition: session
>>>>            Content-ID: <4e5562cd1214427d@example.net>
>>>>
>>>> In addition, what happens to the INVITE request if the 
>> recipient can not get
>>>> the indirect content?
>>>>
>>>> Nits: a letter "l" is missing in the title.
>>>> Internet-Draft         Message Body Hand"l"ing in SIP           
>>   May 2007
>>>> Cheers,
>>>> Qian
>>>>
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Sent: Friday, July 20, 2007 2:30 AM
>>>> To: Gonzalo Camarillo
>>>> Cc: IETF SIP List
>>>> Subject: Re: [Sip] comments on draft-camarillo-sip-body-
>> handling-01
>>>> Just one comment in addition to what Gonzalo said:
>>>>
>>>> Gonzalo Camarillo wrote:
>>>>
>>>>>> The use case described for multipart/alternative - new SDP 
>> mechanisms>>>> - seems kind of bogus. We've decided not to pursue 
>> that. Do we have a
>>>>>> REAL use case for this?
>>>>> AFAICT, we have decided not to pursue SDPng but we still need a
>>>>> mechanism in order to be able to, at some point, migrate to 
>> new session
>>>>> description formats. The current consensus, as I understand 
>> it, is that
>>>>> multipart/alternative should not be used to provide 
>> alternative SDP
>>>>> descriptions, but that it could be used to provide alternative 
>> session>>> descriptions written in different formats.
>>>> The other classic example, because it is just like email, is 
>> MESSAGE>> with a multipart/alternative containing plain text and html.
>>>> Paul
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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>>>
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> 


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From sip-bounces@ietf.org Mon Jul 23 10:22:49 2007
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Hi Dale,

ext sip-bounces@ietf.org wrote:
>    > Often a subscriber will send the SUBSCRIBE to a URI which is not the
>    > request-URI of the SUBSCRIBE when it reaches the notifier agent.  This
>    > happens due to normal SIP request routing, but the routing can also
>    > change with time.
> 
>    Correct, but are you saying there might be more than one agent with a
>    different idea on the entity-tags of entities pertaining to a certain
>    resource? In other words, that the resource has multiple agents that may
>    have a different view of the resource state?
> 
>    I believe this is disallowed per RFC3265 already.
> 
> I doubt things could work with multiple agents that have a different
> view of the resource state.
> 
> But there might be multiple agents with different addresses, from whom
> you request the state of the same resource using different SIP URIs.
> (Which ultimately might all be destinations reached by sending
> requests to the same AOR.)
> 
> I don't think it interferes with the subnot mechanism, but my point is
> that a resource might *not* have a single, specific SIP URI, either as
> the request-URI when the SUBSCRIBE reaches the agent(s) which can
> deliver its state, or as the address to which UAs subscribe.

That's fine. I think aliases work as long as the relationship is clear.
A single resource can have multiple URIs pointing to it, and each
resource will only ever have a single agent responsible for it.

> All you need to make subnot work is that every UA agrees on which
> subscriptions are "for the same resource", and that all notifiers for
> "the same resource" generate the same etags for the same state, and
> different etags for different state.

Right, that would need to be spelled out in section 5.1., I think.

Cheers,
Aki


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From sip-bounces@ietf.org Mon Jul 23 10:27:13 2007
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Sorry for the clutter...=20

Posting to SIP list (stupid auto-complete feature hosed me).

Regards,
Brian

-----Original Message-----
From: Stucker, Brian (RICH1:AR00)=20
Sent: Thursday, July 19, 2007 11:51 PM
To: 'Cullen Jennings'; 'Rohan Mahy'
Cc: sipping@ietf.org; Audet, Francois (SC100:3055)
Subject: Question about keep-alives in Outbound draft

Cullen/Rohan,

I have a question about keep-alive behavior in the -10 version of the
outbound draft.

How far are keep-alives propagated along the flow created at
registration?

The text seems a bit hazy on how far it is expected for a keep-alive to
be propagated along the flow. Section 8 has text in the last paragraph
on page 25 that seems to imply that STUN keep-alives go no further than
the edge proxy immediately adjacent to the client device. Section 4.4.3
seems to also limit generation of STUN keep-alives to the User-Agent
only.

Likewise, section 4.4.1 and 5.4 seem to limit the TCP CRLF response to
the first edge-proxy as well.

Inclusion of the keep-stun or keep-crlf parameters for a Path URI
inserted by an edge proxy or registrar outside of the one in the
outbound-proxy-set seems moot as earlier edge-proxies that don't support
these mechanisms would ostensibly drop the keep-alives.

Further, you only show a NAT/FW between the first edge-proxy and the
User-Agent in the figure for section 3.4. However, the spec never really
says very clearly what should occur at the edge-proxy: should it check
for freshness of the next-hop towards the registrar or not, based upon
what it observed in the Path header coming back in the registration
response?

Is there a reason that edge-proxies are not expected to regenerate/relay
the appropriate keep-alive message to the next entry on the Path vector?
Not doing so means that some other mechanism must be employed to keep
bindings/pinholes open between the various edge-proxies and the
registrar proxy.

For instance, in IMS you can have a P-CSCF in the visited network
talking to an IBCF in the visited network that is connected to an IBCF
in the home network that finally terminates at the S-CSCF in the home
network of the user that is registering. A NAT/FW is likely to exist
between the UA and the visited P-CSCF as well as between the IBCF in the
visited network and the IBCF in the home network.

Another possible example (I believe) would be in the case of p2psip
where you may have multiple hops between the client and the node acting
as a registrar for the DHT algorithm. Wouldn't you want to keep the
entire path to the node storing your contact information in the DHT
alive or is there some other property of the DHT algorithm that ensures
that this is always the case even though outbound only keeps the last
hop fresh?

Regards,
Brian


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From sip-bounces@ietf.org Mon Jul 23 10:27:19 2007
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 Sorry for the clutter...=20

Posting to SIP list (stupid auto-complete feature hosed me).

Regards,
Brian

-----Original Message-----
From: Stucker, Brian (RICH1:AR00)=20
Sent: Friday, July 20, 2007 12:47 AM
To: 'Cullen Jennings'; 'Rohan Mahy'
Cc: 'sipping@ietf.org'; Audet, Francois (SC100:3055);
'cboulton@ubiquitysoftware.com'; Jonathan Rosenberg; Gonzalo Camarillo
Subject: Another comment on the outbound draft

Rohan/Cullen,

One other comment on the outbound draft... I think a statement needs to
be added to the draft to cover the following situation:

Let's say a UAC is registering from behind a NAT. If the outbound proxy
does not support outbound (i.e. does not put 'ob' in the Path header)
then the resulting registration will contain an address that is likely
not reachable by the local outbound proxy after the NAT binding has
expired. The UAC can detect that this may occur by looking for the
'outbound' tag in the supported header of the registration response and
act accordingly. However, in section 4.2 the text in the third paragraph
on page 15 shows that checking the supported header to see if outbound
processing occurred is only a MAY instead of a MUST.

I believe that the UAC must do this checking unless they know for a fact
that their contact address is globally routable (or at least reachable
by the local outbound proxy at all times), and if not, re-register using
their server reflexive address in the contact URI via STUN or derived
through some other means (observation of received/rport values in the
registration response Via header, etc).

Otherwise, the client may have successfully registered an unroutable
address without realizing it thinking that the connection to the
outbound proxy obviated the need to specify a globally routable address
in the contact URI. The BCP for NAT traversal only points off to
outbound (although the citation is very stale in the draft, along with
the even more stale reference to ANAT as a draft, although it has a more
recent ICE reference) and does not discuss this failure case either. In
fact, it directly washes it's hands of it:=20

	[outbound] exonerates the need to manipulate contact header
URI's to represent a globally routable address as 	perceived on the
public side of a NAT.

Regards,
Brian


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Like to point out there are some experimental implications that are  
using the 419 status code for spam stuff and we might no want to use  
that one.



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Subject: RE: [Sip] comments on draft-camarillo-sip-body-handling-01
Date: Mon, 23 Jul 2007 21:45:06 +0200
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "sunqian 32328" <sunqian@huawei.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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They could, but I suspect we would end up in a discussion with the APPS
area of why we are trying to.=20

Either we implement MIME or we don't.=20

But to continue the discussion - if you want to do this, then you are
going to have to motivate it. I don't see us taking a hum per method and
arbitrarily going one way on one method and another way on another.

And I don't see the WG being motivated by prior implementation either. I
believe we are at the point where we need to make a clear statement on
what SIP needs going forwards, not a kludge based on what some existing
implementations require to validate their conformance to the new
requirements.

Regards

Keith

=20

> -----Original Message-----
> From: sunqian 32328 [mailto:sunqian@huawei.com]=20
> Sent: Monday, July 23, 2007 12:03 PM
> To: Paul Kyzivat
> Cc: 'IETF SIP List'; Gonzalo Camarillo
> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
>=20
> Hi,
>=20
> Could different SIP methods (INVITE, MESSAGE) be given=20
> different Levels of Support for Multipart?
>=20
> Best Regards,
> Qian Sun
>=20
> ----- Original Message -----
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Date: Sunday, July 22, 2007 12:21 pm
> Subject: Re: [Sip] comments on draft-camarillo-sip-body-handling-01
>=20
> > One of the tasks of the draft is to make support for at least some=20
> > subtypes of multipart (e.g. mixed) mandatory.
> >=20
> > I hope this is not a suggestion to make dereferencing of external=20
> > bodies mandatory.
> >=20
> > If the request is just to include some clarification of the use and=20
> > handling of external bodies in this draft then I guess that=20
> does sound=20
> > useful.
> >=20
> > 	Paul
> >=20
> > Eric Burger wrote:
> > > Seems like a good idea.  However, I would not go so far as to
> > mandate> iterative evaluation.  I would offer that the UA
> > evaluates as needed.
> > >=20
> > >=20
> > > On 7/20/07 3:58 AM, "Qian Sun" <sunqian@huawei.com> wrote:
> > >=20
> > >> Hi,
> > >> I suggest Content Indirection Mechanism defined in rfc4483
> > should also be
> > >> considered in this draft.
> > >>
> > >> It is a little special, an example from rfc4483:
> > >> 6.1.  Single Content Indirection
> > >>
> > >>            INVITE sip:boromir@example.com SIP/2.0
> > >>            Accept: message/external-body application/sdp
> > >>            Content-Type: message/external-body;
> > >>                 ACCESS-TYPE=3DURL;
> > >>                =20
> > URL=3D"http://www.example.net/party/06/2002/announcement";>>         =

> >        EXPIRATION=3D"Sat, 20 Jun 2002 12:00:00 GMT";
> > >>                 size=3D231
> > >>            Content-Length: 105
> > >>
> > >>            Content-Type: application/sdp
> > >>            Content-Disposition: session
> > >>            Content-ID: <4e5562cd1214427d@example.net>
> > >>
> > >> In addition, what happens to the INVITE request if the
> > recipient can not get
> > >> the indirect content?
> > >>
> > >> Nits: a letter "l" is missing in the title.
> > >> Internet-Draft         Message Body Hand"l"ing in SIP          =20
> >   May 2007
> > >>
> > >> Cheers,
> > >> Qian
> > >>
> > >> -----Original Message-----
> > >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > >> Sent: Friday, July 20, 2007 2:30 AM
> > >> To: Gonzalo Camarillo
> > >> Cc: IETF SIP List
> > >> Subject: Re: [Sip] comments on draft-camarillo-sip-body-
> > handling-01
> > >>
> > >> Just one comment in addition to what Gonzalo said:
> > >>
> > >> Gonzalo Camarillo wrote:
> > >>
> > >>>> The use case described for multipart/alternative - new SDP
> > mechanisms>>>> - seems kind of bogus. We've decided not to pursue
> > that. Do we have a
> > >>>> REAL use case for this?
> > >>> AFAICT, we have decided not to pursue SDPng but we still need a=20
> > >>> mechanism in order to be able to, at some point, migrate to
> > new session
> > >>> description formats. The current consensus, as I understand
> > it, is that
> > >>> multipart/alternative should not be used to provide
> > alternative SDP
> > >>> descriptions, but that it could be used to provide alternative
> > session>>> descriptions written in different formats.
> > >> The other classic example, because it is just like email, is
> > MESSAGE>> with a multipart/alternative containing plain=20
> text and html.
> > >>
> > >> Paul
> > >>
> > >>
> > >> _______________________________________________
> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >> This list is for NEW development of the core SIP Protocol Use=20
> > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >> sipping@ietf.org for new developments on the application of sip
> > >>
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >> This list is for NEW development of the core SIP Protocol Use=20
> > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > >> sipping@ietf.org for new developments on the application of sip
> > >=20
> > >=20
> > > Notice:  This email message, together with any attachments, may
> > contain information  of  BEA Systems,  Inc.,  its subsidiaries and =20
> > affiliated entities,  that may be confidential,  proprietary,=20
> > copyrighted  and/or legally privileged, and is intended=20
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> > please immediately return this by email and then delete it.
> > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: [Sip] ICE tutorial in Chicago is CONFIRMED [DO NOT REPLY] - lunch
 logistics below
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(Apologies to multiple recipients)
(DO NOT REPLY)

WHAT:     Tutorial on Interactive Connectivity Establishment (ICE)
      (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-17.txt)

           Slides:
           http://www.jdrosen.net/papers/ice-ietf-tutorial2.ppt

WHEN:     Tuesday, July 24, 2007, 1145-1245
           lunch will be available to those who RSVPd previously
           sandwiches, chips, salad and drinks
           $20/head - please pay Jonathan R. after the tutorial

WHERE:    Red Lacquar
           MP3 streaming hopefully available at:
           http://videolab.uoregon.edu/events/ietf/ietf694.m3u

WHO:      Anyone that would like to learn more about ICE.

WHY:      ICE, developed by the mmusic working group, has been
           defined as the IETF tool for NAT traversal for SIP-based
           media. It is also seeing increased usage for traversal
           for other kinds of application traffic - anything worrying
           about any-to-any communications. In this tutorial, we will
           overview the background of the problem, consider the
           requirements driving ICE, and provide an overview of
           how ICE works. An understanding of SIP is not required.
           The tutorial will be based on the pending -17 version.

RSVP:     Anyone may attend without RSVP, but lunch may not be
           there for you if you have not RSVPd previously. I've ordered
           a few extra


!DO NOT REPLY TO THIS NOTE!
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com



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At 19:10 21/07/2007, Henry Sinnreich wrote:
>I fully agree with Jiri that this is an IETF problem and not an ICE
>problem.
>ICE-17 is a great document, no doubt about it.
>
>What happens however of someone finds an innovative approach that may
>work better? Will it be rejected because it will not be RFC standards
>compliant?

I don't think so, IETF is having a history of working on competing standards,
jabber and simple being a prominent and recent example thereof. I think that's
actually with current IETF's size and dynamics a fairly good strategy to
eliminate single points of failure. 

-jiri


>What could an innovator tell customers about a solution that works
>better but is not IETF ICE RFC standards compliant?
>
>Henry
>
>-----Original Message-----
>From: Jiri Kuthan [mailto:jiri@iptel.org] 
>Sent: Saturday, July 21, 2007 9:48 AM
>To: David Barrett; 'Jan Janak'; 'Juha Heinanen'
>Cc: sip@ietf.org; mmusic@ietf.org
>Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
>deploymentdatabefore LC for RFC
>
>At 20:05 19/07/2007, David Barrett wrote:
>>> -----Original Message-----
>>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>>> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment
>databefore
>>> LC for RFC
>>> 
>>> So I think that the right course of action is that folks doubting ICE
>>> reliability
>>> make their case (by that I mean specific arguments rather than
>emotional
>>> debates
>>> about red-colored nails and other hardware).
>>
>>Er, my "specific argument" is "until it's been proven to work, it
>hasn't
>>been proven to work".  It's tautological, but no less true for it.
>>
>>Indeed, if you're *not* doubting ICE, why aren't you?
>
>Hi David,
>
>I'm not entirely sure what is the step you are suggesting?
>
>I think ICE is architecturally right answer to the right problem.
>Notwithstanding that, there are certainly some things I prefer
>differently (e.g., the infamous debate about proactive hole punching)
>or may turn out to be bugs. I think though that the key problem at
>this point is of institutional nature. In the IETF language, "hasn't
>been proven to work" is adequate to "lacking running code" and this
>appears to be a phenomen which is indeed occuring in the IETF and
>is not unique to ICE.
>
>The question is how to get this fixed?
>
>Some other standardization bodies enjoying the amount of contributions
>and complexity as the IETF is enjoying now (S in SIP does not really
>stand for Simple), proceed with 'after-fact filters', in that they
>produce the initial specs faster and later produce 'profiles' which
>separate what has proved and what less. Plus -bis versions are produced
>for whatever appears worth puttting effort in it (i.e., for which there
>is running code and field evidence of improvement need).This could be 
>an approach  which may be more feasible in today's IETF's dimension than
>
>trying to get back to historical roots.
>
>-jiri 
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



--
Jiri Kuthan            http://iptel.org/~jiri/



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   From: Aki Niemi <aki.niemi@nokia.com>

   > All you need to make subnot work is that every UA agrees on which
   > subscriptions are "for the same resource", and that all notifiers for
   > "the same resource" generate the same etags for the same state, and
   > different etags for different state.

   Right, that would need to be spelled out in section 5.1., I think.

I probably didn't make myself clear, in that this is just a matter of
wording and terminology.  But if the RFC isn't worded correctly,
people might create non-workable implementations.

Dale


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From sip-bounces@ietf.org Mon Jul 23 18:22:01 2007
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Comments on draft-ietf-sip-outbound-10

by Alfred E. Heggestad <aeh@db.org>
23. July 2007




 > 3.2.  Single Registrar and UA
 >
 >   REGISTER sip:example.com;keep-crlf SIP/2.0
 >   Via: SIP/2.0/TCP 192.0.2.1;branch=z9hG4bK-bad0ce-11-1036
 >   Max-Forwards: 70
 >   From: Bob <sip:bob@example.com>;tag=d879h76
 >   To: Bob <sip:bob@example.com>
 >   Call-ID: 8921348ju72je840.204
 >   CSeq: 1 REGISTER
 >   Supported: path
 >   Contact: <sip:line1@192.168.0.2>; reg-id=1;
 >    ;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000A95A0E128>"
 >   Content-Length: 0

I find this example SIP message a bit strange. If the message is sent
over TCP, shouldn't the URI in the Contact header contain ";transport=tcp" ?
Also, is the IP address in the Via header supposed to be a private IP
address (192.168.x.y) or a public IP address?




 > 3.5.3.  STUN Keepalive Technique
 >
 >   This technique can only be used for transports, such as UDP, that are
 >   not connection oriented.

Suggest rewriting this sentence:

    "This approach MUST only be used with connection-less transports
     such as UDP."




 > 4.4.3.  Keepalive with STUN
 >
 >   This approach MUST only be used with transports, such as UDP, that
 >   are not connection oriented.

same as above




 > 12.2.  SIP/SIPS URI Parameters
 >
 >       Parameter Name  Predefined Values  Reference
 >       ____________________________________________
 >       keep-crlf           No            [RFC AAAA]
 >       keep-stun           No            [RFC AAAA]

suggest to merge these two parameters to "keepalive"




 > 12.2.  SIP/SIPS URI Parameters
 >
 >       Parameter Name  Predefined Values  Reference
 >       ____________________________________________
 >       timed-keepalive     No            [RFC AAAA]
 >       ^^^^^^^^^^^^^^^

This URI parameter is referred to as 'timed-keepalives' (with the trailing 's')
a total of 4 locations in the document. They should be updated to read
"timed-keepalive"




apart from these comments, the document is well structured and easy to read.


/alfred



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From sip-bounces@ietf.org Tue Jul 24 00:23:52 2007
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A reminder of the ongoing WGLC on the above document

This WGLC period runs through IETF and hopefully close August 6, 2007.

That gives you about a week after the end of IETF - please schedule some =
time for doing this.

Regards

Keith
=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Thursday, July 12, 2007 7:14 PM
> To: Dean Willis
> Cc: sip@ietf.org; Cullen Jennings; DRAGE, Keith (Keith)
> Subject: Correction: WGLC for draft-ietf-sip-outbound
>=20
> Dean Willis wrote:
> > I'd like to start a long working group last call for:
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
> >
>=20
> My error. This should have referred to=20
> draft-ietf-sip-outbound-10, not -09. It was busy working its=20
> way through the posting system and I grabbed a pointer to the=20
> wrong one.
>=20
> Sorry about that, folks.
>=20
> --
> Dean
>=20
>=20


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Should there be one more change to 3261 for invfix?

Section 14.1 para 6 point # 2 should include the Accepted state?

"If there is an ongoing INVITE server transaction, the TU MUST wait until
the transaction reaches the confirmed, *accepted* or terminated state before
initiating the new INVITE"

Thanks

Nasir Khan

BEA Systems Inc.

On 7/17/07, Robert Sparks <rjsparks@estacado.net> wrote:
>
>
> On Jul 17, 2007, at 1:42 AM, Nasir Khan wrote:
>
> Hi Robert,
>
> I have 3 comments / suggestions w.r.t this ID.
>
> Comment-1
> ----------------
> Should there be any change with respect to receiving CANCEL in Accepted
> state? Hitherto the UAC was getting a 481/CANCEL in such cases. Now we will
> get 200/CANCEL with no effect on IST.
> The reason for this question is backwards compatibility, however I can
> myself argue both ways on this one. What do you think?
>
>
> I don't see any issue here. The UAC already has to be prepared for either
> response.
>
>
>
>
> Comment-2
> -----------------
> In section 6.7 of the ID about the Timer-L you say - "and the amount of
> time this (or any downstream) UAS core might be retransmitting the 2xx while
> waiting for an ACK. "
>
> How does 2xx retransmission timer affect the choice of Timer L?
>
>
>
> Comment-3
> -----------------
> Shouldnt the Timer L be set to T4 which represents the amount of time the
> network will take to clear messages between client and server transactions?
> Since 2xx is sent reliably anyway, IMHO we do not need to link Timer L to
> Timer B.
>
>
> For both of these comments - Timer L is keeping the transaction in place
> while the TU on top of this server transaction finishes its
> job of retrasmitting 200s until it gets an ACK. It's not just clearing
> messages that are already in the network. Remember that those
> 200s might get lost, and timers on the UAC's side will cause the INVITE
> request to be retransmitted. This Timer makes sure the
> server transaction is still there to receive that retransmitted INVITE.
> Does this clear up the questions you had?
>
>
>
> Thanks
>
> Nasir Khan
> BEA Systems Inc.
>
>
> On 6/27/07, Robert Sparks <rjsparks@estacado.net> wrote:
> >
> > Everyone -
> >
> > I've put together a proposal for addressing the spec bug captured at
> > http://bugs.sipit.net/show_bug.cgi?id=769
> >
> > Please read through
> > http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
> > and comment.
> >
> > RjS
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>
>
> --
> Discuss SIP Servlets at http://groups.google.com/group/sipservlets/
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>


-- 
Discuss SIP Servlets at http://groups.google.com/group/sipservlets/

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Should there be one more change to 3261 for invfix?<br>
<br>
Section 14.1 para 6 point # 2 should include the Accepted state? <br>
<br>
&quot;If there is an ongoing INVITE server transaction, the TU MUST wait
until the transaction reaches the confirmed, *accepted* or terminated
state before initiating the new INVITE&quot;<br>
<br>
Thanks<br>
<br>
Nasir Khan <br>
<br>
BEA Systems Inc. <br>
<br><div><span class="gmail_quote">On 7/17/07, <b class="gmail_sendername">Robert Sparks</b> &lt;<a href="mailto:rjsparks@estacado.net">rjsparks@estacado.net</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style=""><br><div><span class="q"><div>On Jul 17, 2007, at 1:42 AM, Nasir Khan wrote:</div><br><blockquote type="cite">Hi Robert,<br> <br> I have 3 comments / suggestions w.r.t this ID. <br> <br> Comment-1<br> ----------------
<br>
Should there be any change with respect to receiving CANCEL in Accepted
state? Hitherto the UAC was getting a 481/CANCEL in such cases. Now we
will get 200/CANCEL with no effect on IST. <br> The reason for this question is backwards compatibility, however I can myself argue both ways on this one. What do you think?<br></blockquote><div><br></div></span>I don&#39;t see any issue here. The UAC already has to be prepared for either response.&nbsp;
</div><span class="q"><div>&nbsp;<br><blockquote type="cite"> <br> <br> Comment-2<br> -----------------<br>
In section 6.7 of the ID about the Timer-L you say - &quot;and the amount of
time this (or any downstream) UAS core might be retransmitting the 2xx
while waiting for an ACK. &quot; <br> <br> How does 2xx retransmission timer affect the choice of Timer L?&nbsp;</blockquote></div></span><div><span class="q"><blockquote type="cite"> <br> <br> Comment-3<br> -----------------
<br>
Shouldnt the Timer L be set to T4 which represents the amount of time
the network will take to clear messages between client and server
transactions? <br> Since 2xx is sent reliably anyway, IMHO we do not need to link Timer L to Timer B.<br></blockquote><div><br></div></span>For
both of these comments - Timer L is keeping the transaction in place
while the TU on top of this server transaction finishes its</div><div>job
of retrasmitting 200s until it gets an ACK. It&#39;s not just clearing
messages that are already in the network. Remember that those</div><div>200s might get lost, and timers on the UAC&#39;s side will cause the INVITE request to be retransmitted. This Timer makes sure the</div><div>server transaction is still there to receive that retransmitted INVITE.&nbsp; Does this clear up the questions you had?
</div><div><br></div><div><br><blockquote type="cite"><span class="q"> <br> Thanks<br> <br> Nasir Khan<br> BEA Systems Inc.<br> <br><br><div><span class="gmail_quote">On 6/27/07, <b class="gmail_sendername">Robert Sparks</b>
 &lt;<a href="mailto:rjsparks@estacado.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">rjsparks@estacado.net</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
 Everyone -<br><br>I&#39;ve put together a proposal for addressing the spec bug captured at<br><a href="http://bugs.sipit.net/show_bug.cgi?id=769" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://bugs.sipit.net/show_bug.cgi?id=769
</a><br><br>Please read through<br> <a href="http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
</a><br>and comment.<br><br>RjS<br><br><br>_______________________________________________ <br>Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a href="mailto:sip-implementors@cs.columbia.edu" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
 sip-implementors@cs.columbia.edu</a> for questions on current sip<br>Use <a href="mailto:sipping@ietf.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sipping@ietf.org</a> for new developments on the application of sip
<br></blockquote></div><br><br clear="all"><br> -- <br>Discuss SIP Servlets at <a href="http://groups.google.com/group/sipservlets/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://groups.google.com/group/sipservlets/
</a></span><div style="margin: 0px;">_______________________________________________</div><span class="q"><div style="margin: 0px;">Sip mailing list<span>&nbsp; </span><a href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
https://www1.ietf.org/mailman/listinfo/sip</a></div><div style="margin: 0px;">This list is for NEW development of the core SIP Protocol</div><div style="margin: 0px;">Use <a href="mailto:sip-implementors@cs.columbia.edu" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
sip-implementors@cs.columbia.edu</a> for questions on current sip</div><div style="margin: 0px;">Use <a href="mailto:sipping@ietf.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sipping@ietf.org
</a> for new developments on the application of sip</div> </span></blockquote></div><br></div></blockquote></div><br><br clear="all"><br>-- <br>Discuss SIP Servlets at <a href="http://groups.google.com/group/sipservlets/">
http://groups.google.com/group/sipservlets/</a>

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From: "Xavier Marjou" <xavier.marjou@orange-ftgroup.com>
To: "Eric Burger" <eburger@bea.com>
Subject: Re: [Sip] INFO and Content Negotiation
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Hi Eric,

I also do not understand why it would not be possible to indicate in
the transaction of the initial INVITE what type of INFO is exactly
required/supported (negotiation and subscription stage) and to send
the event information with an INFO during the dialog (notification
stage).

Concerning draft-burger-sip-info-00, I think the draft is not fair
with this poor INFO message.  Indeed, the INFO message helps a lot to
quickly implement/have some features (e.g. full intra requests of
mmusic-xml-media-control, DTMF, MSCML, ...) while waiting for
implementations of the IETF standardized solutions (e.g.
rtcp-feedback, kpml, media-control, ...). Though I agree there are
some cons with features based on INFOs messages, there are also some
pros with this approach, (ie: same as for everything ;-) but I won't
go over such details. Thus, I think that even if the IETF does not
want to recommend the INFO message, the IETF should at least recognize
it for experimental uses (i.e. roughly an equivalent of RTCP APP
packet) and even documents these uses. Otherwise, we can end up with
some types of INFO messages that are broadly used but that are even
not documented within IETF documents (e.g. application/dtmf-relay for
INFO with DTMF)

Xavier



On 7/20/07, Eric Burger <eburger@bea.com> wrote:
> 12 second proof point of why content type doesn't work.
>
> So here comes an INVITE.  I am an INFO-friendly UAS.  I negotiate by using
> Accept.
>
> This INVITE has the line:
> Accept: application/kpml-response+xml
>
>
> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?  Because it accepts
> a DTMF content type.  Excellent - I can now send them DTMF.
>
> The user enters a digit, and I send an INFO with type
> application/kpml-response+xml.
>
> UAC says, "What the heck was that?"
>
>
> Notice:  This email message, together with any attachments, may contain information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entities,  that may be confidential,  proprietary,  copyrighted  and/or legally privileged, and is intended solely for the use of the individual or entity named in this message. If you are not the intended recipient, and have received this message in error, please immediately return this by email and then delete it.
>
>
> _______________________________________________
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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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Overall this draft is excellent and exactly what I think it should be 
for this. Some comments below, major and nits:

Major:


Section 1: There are some additional problems worth noting:
   * privacy service has to be a b2bua and thus interferes with security 
mechanisms that break in the face of b2bua, such as sip identity (which 
provides message integrity and works well here too to provide the remote 
side an assurance of authenticated identity, just one that is not 
provided), which in turn breaks dtls-srtp

* because of the need to signal from the UA to the privacy service 
exactly the desired functionality, the set of functionalities possible, 
in terms of what is anonyzmized and what is not, is bounded and required 
detailed specification.


> The user agent setting the "temp-gruu" as a GRUU SHOULD set
>    "Anonymous" as a display name in any header where the display name of
>    the originator is set.  That indicates the anonymity of the request
>    to intermediaries that may invoke some services based on the
>    anonymity of the call.

Hmm, I'd prefer something explicit. There are problems with using 
human-readable and locale-specific text like display name to signal to 
automata.


> Note: A relayed IP address may be used for a Via header, but some
>          commented that is not an appropriate to be used for signaling.
>          There was a comment about the IP address in Via being stripped
>          by the proxy, but that would require that a proxy compliant to
>          this specification is in the signaling path.

I think it was me you said you didn't need this. You make a good point 
here however.

> Note: The mechanism of the flag is FFS.

 From text above I thought it was presence of the Anonymous in display 
name, which is bad. I personally like the Privacy: id value since its 
effectively what everyone is using and also backwards compatible.

I also think we need something to make sure the terminating UA doesn't 
try to render the From or P-A-ID if someone should insert.

>  Proxy Behavior

The draft should talk about RFC 3325 in particular.

Nits:


> privacy service.  This document proposes a new privacy mechanism that
>    a user agent can facilitate to conceal privacy-sensitive information
>    without the need for aid from a privacy service.

This is somehwat misleading; there is a privacy service in the form of 
turn servers and proxies.

> The protection of network
>    privacy (e.g., topology hiding) is outside the scope of this
>    document.

You need to clarify what this means.

> A user agent wanting to obtain functional anonymous URI SHOULD
>    support and SHOULD utilize the Global Routable User Agent URI (GRUU)
>    mechanism. 

Isn't this going to need to be a MUST?

> response for a "temp-gruu" URI parameter, which provides the desired
>    privacy property.
provides URI with the desired property

> It is assumed that a user agent is either manually or automatically
>    configured through means such as a configuration framework with one
>    or more STUN relay servers.

terminology has changed again; these are now called TURN servers once more.



Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org Tue Jul 24 08:17:26 2007
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Date: Tue, 24 Jul 2007 07:16:28 -0500
Subject: Re: [Sip] INFO and Content Negotiation
From: Eric Burger <eburger@bea.com>
To: Xavier Marjou <xavier.marjou@orange-ftgroup.com>
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Indicating the package is possible; it is what Dean's draft of 4 years ago
proposed.  However, it is not what is there today.  Moreover, it is doubtful
that any of the myriad INFO exploits would use the mechanism.  If one is
going to upgrade one's mechanism, one might as well do the right thing.

I would also offer we learn the lesson from the e-mail world of X-mumble
headers.

It is true that we need a mechanism to experiment with new extensions to
SIP.  However, formalizing that mechanism is asking for a rash of
incompatible half-baked protocols, some of which, while broken, hijack SIP.

The INFO draft as it currently stands still "allows" for such
experimentation with INFO.  It sounds like if we do want to work on this
draft, the language needs to be clearer that (1) INFO is not gone and (2)
INFO is *not* how you deploy something outside of the lab.

Just for those who may have missed it, Section 4 states:
   In recognition of existing, proprietary use of INFO, proxies MUST NOT
   take any action other than that described by RFC 3261 and RFC 2976
   with respect to handling INFO requests.

This is the escape clause.


On 7/23/07 11:45 PM, "Xavier Marjou" <xavier.marjou@orange-ftgroup.com>
wrote:

> Hi Eric,
> 
> I also do not understand why it would not be possible to indicate in
> the transaction of the initial INVITE what type of INFO is exactly
> required/supported (negotiation and subscription stage) and to send
> the event information with an INFO during the dialog (notification
> stage).
> 
> Concerning draft-burger-sip-info-00, I think the draft is not fair
> with this poor INFO message.  Indeed, the INFO message helps a lot to
> quickly implement/have some features (e.g. full intra requests of
> mmusic-xml-media-control, DTMF, MSCML, ...) while waiting for
> implementations of the IETF standardized solutions (e.g.
> rtcp-feedback, kpml, media-control, ...). Though I agree there are
> some cons with features based on INFOs messages, there are also some
> pros with this approach, (ie: same as for everything ;-) but I won't
> go over such details. Thus, I think that even if the IETF does not
> want to recommend the INFO message, the IETF should at least recognize
> it for experimental uses (i.e. roughly an equivalent of RTCP APP
> packet) and even documents these uses. Otherwise, we can end up with
> some types of INFO messages that are broadly used but that are even
> not documented within IETF documents (e.g. application/dtmf-relay for
> INFO with DTMF)
> 
> Xavier
> 
> 
> 
> On 7/20/07, Eric Burger <eburger@bea.com> wrote:
>> 12 second proof point of why content type doesn't work.
>> 
>> So here comes an INVITE.  I am an INFO-friendly UAS.  I negotiate by using
>> Accept.
>> 
>> This INVITE has the line:
>> Accept: application/kpml-response+xml
>> 
>> 
>> Wow!  This is great!  I know the UAC *wants* DTMF.  Why?  Because it accepts
>> a DTMF content type.  Excellent - I can now send them DTMF.
>> 
>> The user enters a digit, and I send an INFO with type
>> application/kpml-response+xml.
>> 
>> UAC says, "What the heck was that?"
>> 
>> 
>> Notice:  This email message, together with any attachments, may contain
>> information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated
>> entities,  that may be confidential,  proprietary,  copyrighted  and/or
>> legally privileged, and is intended solely for the use of the individual or
>> entity named in this message. If you are not the intended recipient, and have
>> received this message in error, please immediately return this by email and
>> then delete it.
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 
> 



Notice:  This email message, together with any attachments, may contain information  of  BEA Systems,  Inc.,  its subsidiaries  and  affiliated entities,  that may be confidential,  proprietary,  copyrighted  and/or legally privileged, and is intended solely for the use of the individual or entity named in this message. If you are not the intended recipient, and have received this message in error, please immediately return this by email and then delete it.


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From sip-bounces@ietf.org Tue Jul 24 09:17:02 2007
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Subject: Re: [Sip] RE: Correction: WGLC for draft-ietf-sip-outbound
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Keith/Rohan -

There's a moderate list of changes we already know are going to be  
made to the document.

Everyone who wasn't in Chicago's Monday SIP meeting should look at
http://www3.ietf.org/proceedings/07jul/slides/sip-3.pdf for the  
bigger ones.
Rohan has indicated a few smaller ones on list.

Would it be possible to get a list of the known expected changes in  
one place?

RjS

On Jul 23, 2007, at 11:23 PM, DRAGE, Keith ((Keith)) wrote:

> A reminder of the ongoing WGLC on the above document
>
> This WGLC period runs through IETF and hopefully close August 6, 2007.
>
> That gives you about a week after the end of IETF - please schedule  
> some time for doing this.
>
> Regards
>
> Keith
>
>
>> -----Original Message-----
>> From: Dean Willis [mailto:dean.willis@softarmor.com]
>> Sent: Thursday, July 12, 2007 7:14 PM
>> To: Dean Willis
>> Cc: sip@ietf.org; Cullen Jennings; DRAGE, Keith (Keith)
>> Subject: Correction: WGLC for draft-ietf-sip-outbound
>>
>> Dean Willis wrote:
>>> I'd like to start a long working group last call for:
>>>
>>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
>>>
>>
>> My error. This should have referred to
>> draft-ietf-sip-outbound-10, not -09. It was busy working its
>> way through the posting system and I grabbed a pointer to the
>> wrong one.
>>
>> Sorry about that, folks.
>>
>> --
>> Dean
>>
>>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: [Sip] Changes to the INVITE transaction to address bug 769
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Yes - thanks! I'll comb for related text.

RjS

On Jul 23, 2007, at 11:32 PM, Nasir Khan wrote:

> Should there be one more change to 3261 for invfix?
>
> Section 14.1 para 6 point # 2 should include the Accepted state?
>
> "If there is an ongoing INVITE server transaction, the TU MUST wait  
> until the transaction reaches the confirmed, *accepted* or  
> terminated state before initiating the new INVITE"
>
> Thanks
>
> Nasir Khan
>
> BEA Systems Inc.
>
> On 7/17/07, Robert Sparks <rjsparks@estacado.net> wrote:
>
> On Jul 17, 2007, at 1:42 AM, Nasir Khan wrote:
>
>> Hi Robert,
>>
>> I have 3 comments / suggestions w.r.t this ID.
>>
>> Comment-1
>> ----------------
>> Should there be any change with respect to receiving CANCEL in  
>> Accepted state? Hitherto the UAC was getting a 481/CANCEL in such  
>> cases. Now we will get 200/CANCEL with no effect on IST.
>> The reason for this question is backwards compatibility, however I  
>> can myself argue both ways on this one. What do you think?
>
> I don't see any issue here. The UAC already has to be prepared for  
> either response.
>
>>
>>
>> Comment-2
>> -----------------
>> In section 6.7 of the ID about the Timer-L you say - "and the  
>> amount of time this (or any downstream) UAS core might be  
>> retransmitting the 2xx while waiting for an ACK. "
>>
>> How does 2xx retransmission timer affect the choice of Timer L?
>>
>>
>> Comment-3
>> -----------------
>> Shouldnt the Timer L be set to T4 which represents the amount of  
>> time the network will take to clear messages between client and  
>> server transactions?
>> Since 2xx is sent reliably anyway, IMHO we do not need to link  
>> Timer L to Timer B.
>
> For both of these comments - Timer L is keeping the transaction in  
> place while the TU on top of this server transaction finishes its
> job of retrasmitting 200s until it gets an ACK. It's not just  
> clearing messages that are already in the network. Remember that those
> 200s might get lost, and timers on the UAC's side will cause the  
> INVITE request to be retransmitted. This Timer makes sure the
> server transaction is still there to receive that retransmitted  
> INVITE.  Does this clear up the questions you had?
>
>
>>
>> Thanks
>>
>> Nasir Khan
>> BEA Systems Inc.
>>
>>
>> On 6/27/07, Robert Sparks <rjsparks@estacado.net> wrote:
>> Everyone -
>>
>> I've put together a proposal for addressing the spec bug captured at
>> http://bugs.sipit.net/show_bug.cgi?id=769
>>
>> Please read through
>> http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt
>> and comment.
>>
>> RjS
>>
>>
>> _______________________________________________
>> Sip mailing list   https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>
>>
>> -- 
>> Discuss SIP Servlets at http://groups.google.com/group/sipservlets/
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
> -- 
> Discuss SIP Servlets at http://groups.google.com/group/sipservlets/
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">Yes - thanks! I'll comb for =
related text.<DIV><BR =
class=3D"khtml-block-placeholder"></DIV><DIV>RjS</DIV><DIV><BR><DIV><DIV>O=
n Jul 23, 2007, at 11:32 PM, Nasir Khan wrote:</DIV><BR =
class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"cite">Should =
there be one more change to 3261 for invfix?<BR> <BR> Section 14.1 para =
6 point # 2 should include the Accepted state? <BR> <BR> "If there is an =
ongoing INVITE server transaction, the TU MUST wait until the =
transaction reaches the confirmed, *accepted* or terminated state before =
initiating the new INVITE"<BR> <BR> Thanks<BR> <BR> Nasir Khan <BR> <BR> =
BEA Systems Inc. <BR> <BR><DIV><SPAN class=3D"gmail_quote">On 7/17/07, =
<B class=3D"gmail_sendername">Robert Sparks</B> &lt;<A =
href=3D"mailto:rjsparks@estacado.net">rjsparks@estacado.net</A>&gt; =
wrote:</SPAN><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 style=3D""><BR><DIV><SPAN class=3D"q"><DIV>On Jul 17, 2007, =
at 1:42 AM, Nasir Khan wrote:</DIV><BR><BLOCKQUOTE type=3D"cite">Hi =
Robert,<BR> <BR> I have 3 comments / suggestions w.r.t this ID. <BR> =
<BR> Comment-1<BR> ---------------- <BR> Should there be any change with =
respect to receiving CANCEL in Accepted state? Hitherto the UAC was =
getting a 481/CANCEL in such cases. Now we will get 200/CANCEL with no =
effect on IST. <BR> The reason for this question is backwards =
compatibility, however I can myself argue both ways on this one. What do =
you think?<BR></BLOCKQUOTE><DIV><BR></DIV></SPAN>I don't see any issue =
here. The UAC already has to be prepared for either response.=A0 =
</DIV><SPAN class=3D"q"><DIV>=A0<BR><BLOCKQUOTE type=3D"cite"> <BR> <BR> =
Comment-2<BR> -----------------<BR> In section 6.7 of the ID about the =
Timer-L you say - "and the amount of time this (or any downstream) UAS =
core might be retransmitting the 2xx while waiting for an ACK. " <BR> =
<BR> How does 2xx retransmission timer affect the choice of Timer =
L?=A0</BLOCKQUOTE></DIV></SPAN><DIV><SPAN class=3D"q"><BLOCKQUOTE =
type=3D"cite"> <BR> <BR> Comment-3<BR> ----------------- <BR> Shouldnt =
the Timer L be set to T4 which represents the amount of time the network =
will take to clear messages between client and server transactions? <BR> =
Since 2xx is sent reliably anyway, IMHO we do not need to link Timer L =
to Timer B.<BR></BLOCKQUOTE><DIV><BR></DIV></SPAN>For both of these =
comments - Timer L is keeping the transaction in place while the TU on =
top of this server transaction finishes its</DIV><DIV>job of =
retrasmitting 200s until it gets an ACK. It's not just clearing messages =
that are already in the network. Remember that those</DIV><DIV>200s =
might get lost, and timers on the UAC's side will cause the INVITE =
request to be retransmitted. This Timer makes sure the</DIV><DIV>server =
transaction is still there to receive that retransmitted INVITE.=A0 Does =
this clear up the questions you had? =
</DIV><DIV><BR></DIV><DIV><BR><BLOCKQUOTE type=3D"cite"><SPAN class=3D"q">=
 <BR> Thanks<BR> <BR> Nasir Khan<BR> BEA Systems Inc.<BR> =
<BR><BR><DIV><SPAN class=3D"gmail_quote">On 6/27/07, <B =
class=3D"gmail_sendername">Robert Sparks</B> &lt;<A =
href=3D"mailto:rjsparks@estacado.net" target=3D"_blank" onclick=3D"return =
top.js.OpenExtLink(window,event,this)">rjsparks@estacado.net</A>&gt; =
wrote:</SPAN><BLOCKQUOTE class=3D"gmail_quote" style=3D"border-left: 1px =
solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: =
1ex;"> Everyone -<BR><BR>I've put together a proposal for addressing the =
spec bug captured at<BR><A =
href=3D"http://bugs.sipit.net/show_bug.cgi?id=3D769" target=3D"_blank" =
onclick=3D"return =
top.js.OpenExtLink(window,event,this)">http://bugs.sipit.net/show_bug.cgi?=
id=3D769 </A><BR><BR>Please read through<BR> <A =
href=3D"http://www.ietf.org/internet-drafts/draft-sparks-sip-invfix-00.txt=
" target=3D"_blank" onclick=3D"return =
top.js.OpenExtLink(window,event,this)">http://www.ietf.org/internet-drafts=
/draft-sparks-sip-invfix-00.txt </A><BR>and =
comment.<BR><BR>RjS<BR><BR><BR>___________________________________________=
____ <BR>Sip mailing list=A0=A0<A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip" target=3D"_blank" =
onclick=3D"return top.js.OpenExtLink(window,event,this)"> =
https://www1.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" target=3D"_blank" =
onclick=3D"return top.js.OpenExtLink(window,event,this)"> =
sip-implementors@cs.columbia.edu</A> for questions on current sip<BR>Use =
<A href=3D"mailto:sipping@ietf.org" target=3D"_blank" onclick=3D"return =
top.js.OpenExtLink(window,event,this)">sipping@ietf.org</A> for new =
developments on the application of sip <BR></BLOCKQUOTE></DIV><BR><BR =
clear=3D"all"><BR> -- <BR>Discuss SIP Servlets at <A =
href=3D"http://groups.google.com/group/sipservlets/" target=3D"_blank" =
onclick=3D"return =
top.js.OpenExtLink(window,event,this)">http://groups.google.com/group/sips=
ervlets/ </A></SPAN><DIV style=3D"margin: =
0px;">_______________________________________________</DIV><SPAN =
class=3D"q"><DIV style=3D"margin: 0px;">Sip mailing list<SPAN>=A0 =
</SPAN><A href=3D"https://www1.ietf.org/mailman/listinfo/sip" =
target=3D"_blank" onclick=3D"return =
top.js.OpenExtLink(window,event,this)"> =
https://www1.ietf.org/mailman/listinfo/sip</A></DIV><DIV style=3D"margin: =
0px;">This list is for NEW development of the core SIP =
Protocol</DIV><DIV style=3D"margin: 0px;">Use <A =
href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_blank" =
onclick=3D"return top.js.OpenExtLink(window,event,this)"> =
sip-implementors@cs.columbia.edu</A> for questions on current =
sip</DIV><DIV style=3D"margin: 0px;">Use <A =
href=3D"mailto:sipping@ietf.org" target=3D"_blank" onclick=3D"return =
top.js.OpenExtLink(window,event,this)">sipping@ietf.org </A> for new =
developments on the application of sip</DIV> =
</SPAN></BLOCKQUOTE></DIV><BR></DIV></BLOCKQUOTE></DIV><BR><BR =
clear=3D"all"><BR>-- <BR>Discuss SIP Servlets at <A =
href=3D"http://groups.google.com/group/sipservlets/"> =
http://groups.google.com/group/sipservlets/</A><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">_______________________________________________</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sip mailing list<SPAN =
class=3D"Apple-converted-space">=A0 </SPAN><A =
href=3D"https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/=
mailman/listinfo/sip</A></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">This list is =
for NEW development of the core SIP Protocol</DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Use <A =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colum=
bia.edu</A> for questions on current sip</DIV><DIV style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Use <A =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</A> for new =
developments on the application of sip</DIV> =
</BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>=

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Date: Tue, 24 Jul 2007 10:36:54 -0400
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Sip] comments on draft-ietf-sip-subnot-etags-00
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Overall I think this document is in great shape. Some comments below, 
mostly minor:

> Changes in connectivity represent another impetus for a subscriber
>    re-subscribing.  If the subscriber's point of attachment to the
>    Internet changes, e.g., due to dynamic address allocation, the
>    subscriber needs to re-subscribe in order to update the dialog
>    endpoint, which is carried in the Contact header field of the
>    SUBSCRIBE request.
> 
>       Another option for reducing connectivity induced subscription
>       refreshes is to use the Globally Routable User Agent (UA) URI
>       (GRUU) [10] as a stable endpoint contact for subscriptions.

Yes, exactly. And here it works really well. So I would remove this 
paragraph as a motivation.

> The same problem affects fetching and polling of resource state as
>    well.  As a benchmark, if we look at the performance of HTTP [9] in
>    similar scenarios, it performs substantially better using conditional
>    requests.

I'm curious if you have actual data on this, or just speculation. most 
HTTP resources are dynamic so its not clear it actually helps http.

> NOTIFY request.  The entity tag is unique for that particular
>    resource and event notification.

You should clarify what it means to be unique for a resource *and* event 
notification. So this means if I fetch the resource with http it has a 
differnet entity tag?

> The two types of conditions available for a SUBSCRIBE are a Suppress-
>    Body-If-Match and a Suppress-Notify-If-Match; each of these header
>    fields contains the last entity-tag seen by the subscriber.  If the
>    condition evaluates to true, the first of these conditions will
>    instruct the notifier to suppress the body of the next notification
>    and the second the entire notification. 

Is there a reason we need both? Could we just have 
Suppress-Body-If-Match and if the subscription state is anything besides 
active and the subscription duration anything different than the Expires 
in the request, a NOTIFY is sent without a body?

> There is also a wildcard entity-tag with
>    a special value of "*" that always matches.

what is the use-case for this?

* I think you can actually implement this without option tags. The 
presence of the Suppress-* header fields basically plays that role.
Supported is needed if the UA is not asking for some feature in a 
request, but merely saying, its supported.

> uppress-Body-If-Match:  if this condition evaluates to true, the
>       notifier is instructed to suppress (i.e., remove) the body of the
>       first NOTIFY request following the SUBSCRIBE.  If false, the
>       notifier follows the default behavior.

Its not 'first' - its any subsequet NOTIFY whose body, had this 
specification not be in use, would contain the resource with that entity 
tag.

> Suppress-Notify-If-Match:  if this condition evaluates to true, the
>       notifier is instructed to suppress (i.e., block) the entire first
>       NOTIFY request following the SUBSCRIBE, and instead send a 204 (No
>       Notification) response.  If false, the notifier again follows the
>       default behavior.

I don't think we need a new response code. What is the justification?

>  The conditional header field can also be wildcarded using the special
>    "*" entity-tag value.  Such a condition always evaluates to true
>    regardless of the value of the current entity-tag for the resource.
> 
>    Example:
> 
>       Suppress-Notify-If-Match: *

This seems like a bad idea in fact. Why is it here?


* Section 4 needs to cover handling of the response to SUBSCRIBE (what 
happens when a 204 comes), and NOTIFY (store entity tag). Also needs to 
handle what to do in the case where the notifier doesn't support the 
mechanism.

> A notifier MUST generate entity tags for each resource it is
>    responsible for.
> 
>       The views might correspond to different groups of users that have
>       varying levels of access rights to the resource state, or to
>       subscribers that have modified their subscription using event
>       notification filtering [11].  For example, in presence [6]
>       watchers may get different levels of accuracy in geolocation
>       information, based on the presentity's privacy settings.

What do views have to do with this?



Nits:



> These procedures have a serious
>    deficiency in that they do not allow state to persist over a
>    subscription refresh, or between two consecutive polls. 

State can persist at the UA; its just that you get notified of stuff you 
already know.

>  An entity-tag is a token carried in the SIP-ETag header field, and it
>    is opaque to the client.  The notifier is free to decide the means
>    for generating an entity-tag, except for the special "*" value.

I think you mean, "The notifier is free to decide on any means for 
generating the entity-tag. It can have any value, except for "*""


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From sip-bounces@ietf.org Tue Jul 24 10:48:41 2007
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Subject: [Sip] Two minor Max-Breadth comments
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During today's meeting, a point was raised surrounding whether the 
max-breadth approach should limit the total number of requests or merely 
the number of requests in flight at once. The statement is that limiting 
only the ones in flight simultaneously ultimately results in the same 
number of messages as would result without the extension.

While there has been a token mention of Timer C, I think we're 
underestimating the mitigating effect of Timer C.

As an additional point, I think it's worth pointing out that the victim, 
if they recognize they are being subjected to this style of attack, can 
tear the entire tree down fairly rapidly by throwing a 600-class response.

/a


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From sip-bounces@ietf.org Tue Jul 24 11:00:43 2007
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   From: Adam Roach <adam@nostrum.com>

   During today's meeting, a point was raised surrounding whether the 
   max-breadth approach should limit the total number of requests or merely 
   the number of requests in flight at once. The statement is that limiting 
   only the ones in flight simultaneously ultimately results in the same 
   number of messages as would result without the extension.

   While there has been a token mention of Timer C, I think we're 
   underestimating the mitigating effect of Timer C.

   As an additional point, I think it's worth pointing out that the victim, 
   if they recognize they are being subjected to this style of attack, can 
   tear the entire tree down fairly rapidly by throwing a 600-class response.

Also, the UAC, if it is not perverse, will ultimately cancel the
request.

Dale


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From sip-bounces@ietf.org Tue Jul 24 11:01:58 2007
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Date: Tue, 24 Jul 2007 09:58:34 -0500
From: Spencer Dawkins <spencer@mcsr-labs.org>
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Subject: [Sip] Notes on Idraft-burger-sip-info-00
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I finally got to look at draft-burger-sip-info-00. I like what I see. I did
notice some nits, and am willing to provide a more formal review, but I 
support adoption of this draft as a WG deliverable, even in its current 
form.

You may have noticed that the MEDIACTRL WG editors were forced to discuss 
INFO in their specifications, because we currently have one draft providing 
guidance on INFO that people seem to generally agree with, but it expired 
years ago. I

Thanks, Eric, for taking this on.

Spencer

detailed comments...

     Intended status: Standards Track

Could/should this be a BCP? Informational isn't strong enough, but...

2.  Flaws With INFO

I'm remembering one other painful point - that INFO didn't provide a way to 
specify content-disposition if there is more than one reasonable choice. Did 
I make this up?

   An interesting issue is every INFO request traverses the same proxy
   path as any other dialog-related SIP request.  Proxies in the path
   that have no interest in INFO requests still must process the
   request.  This may put undue load on those proxies.  What makes this
   issue interesting, and not necessarily a proxy, is one may wish the

Not sure what "and not necessarily a proxy" was supposed to actually be - 
edit error?

   request to traverse the proxy.  The problem is there is no way for
   proxies to not whether or not they have an interest in INFO requests.
   Getting the requests is an all-or-nothing proposition, driven by
   Record-Route.

3.  INFO Alternatives

   What if you think you need UA-to-UA application session signaling?

This sounds like an invitation to therapy :D. Perhaps "If your new 
application requires UA-to-UA session level signaling, what alternatives 
exist that provide this without using INFO?"

   There are four broad classes of session signaling available.  The
   choice depends on the circumstances.
   o  State updates
   o  User stimulus
   o  Direct signaling channel
   o  Proxy-aware signaling

3.1.  State Updates

   This is the broad class of one User Agent updating another with
   changes in state.  Clearly, state updates are the provenance of the
   SUBSCRIBE/NOTIFY [8] event framework.

Excellent vocabulary. Perhaps "Clearly the SUBSCRIBE/NOTIFY [8] event 
framework was specifically designed to meet this need" might be more 
accessible?

3.2.  User Stimulus: Touch Tones and Others

   This is the class of the user entering stimulus at one User Agent,
   and the User Agent transporting that stimulus to the other.  A key
   thing to realize is key presses on the telephone keypad is user
   stimulus.  Thus, the appropriate mechanism to use here is KPML [9].

3.3.  Direct Signaling Channel

   State updates and user stimulus tend to have relatively few messages
   per session.  Sometimes, User Agents have a need for exchanging a

s/have a need for exchanging/need to exchange/

   relatively high number of messages.  In addition, User Agents may
   have a need for a relatively low-latency exchange of messages.  In
   this latter case, the User Agent may not be able to tolerate the
   latency introduced by intermediate proxies.  Likewise, the
   intermediate proxies may have no interest in processing all of that
   data.

... and it's not obvious how they would process it anyway, if the INFO 
payload is really only meaningful to UAs.

   In this case, establishing a separate, direct control channel, as in
   MSRP [10] or MRCPv2 [11] is appropriate.

3.4.  Proxy-Aware Signaling

   Sometimes, one does want proxies to be in the signaling path for UA-
   to-UA application signaling.  In this case, the use of a SIP request
   is appropriate.  To date, there are no mechanisms for completely
   disambiguating INFO requests.  For example, one could create a
   registry of INFO packages.  The definition of the package would
   define the contexts for the various MIME Content-Types, as well as
   the context of the request itself.  However, a package can have
   multiple content types.  Moreover, having the context, or package
   identifier, at the SIP level precludes bundling multiple contexts
   responding in the same INFO request.  For example, a User Agent might
   want to bundle two different responses in a multipart/mixed MIME body
   type.

   Because there is no difference in either the protocol machinery or
   registration process due to these factors, we will not create an INFO
   framework.  If one needs a SIP User Agent-to-SIP User Agent
   application session signaling transport protocol that touches all
   Record-Route proxies in a path, one MUST create a new SIP method as
   described in Section 27.4 of RFC 3261 [2].

4.  INFO Use Clarification

   There is no way to unambiguously use the INFO request in a general
   framework.  The IETF has already standardized use of INFO for SIP-T
   [4].  Thus we will not deprecate the use of INFO for that purpose.
   However, this document explicitly updates INFO [3], in that one MUST
   NOT use the INFO request for anything other than the use described by
   SIP-T.

I'm confused here. Is it "one MUST NOT use INFO" or that "new applications 
MUST NOT define new uses for INFO"?

   In recognition of existing, proprietary use of INFO, proxies MUST NOT
   take any action other than that described by RFC 3261 and RFC 2976
   with respect to handling INFO requests.

      OPEN ISSUE: Do we bow to reality, and say, "INFO is the Port 80 of
      the 2000's.  SBC's will never keep up with newly minted SIP method
      requests, so we keep INFO so we can have a poliferation of
      protocols tunneled over SIP?"

Well, I think your point is actually that SIP is the Port 80 of the 2000s, 
and INFO is the port 80 of SIP signaling, so the situation is even more 
frustrating than your text suggests. My preference is to remove this OPEN 
ISSUE text and publish the draft so we will all drop INFO as a choice for 
future applications.

Appendix A.  Acknowledgements

   Standing on the shoulders of giants.  Jonathan Rosenberg did the
   original "INFO Considered Harmful" on 26 December 2002, which
   influenced the work group and this document.  Likewise, Dean Willis
   influenced the text from his "Packaging and Negotiation of INFO
   Methods for the Session Initiation Protocol" of 15 January 2003.  My,
   we have been working on this for a long time!

Could you mention that these were both Internet-Drafts (ex. "in an 
Internet-Draft submitted on 26 December 2002")? 




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From sip-bounces@ietf.org Tue Jul 24 11:08:29 2007
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Subject: [Sip] Max-Breadth options
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One difficulty with Max-Breadth or any similar feature is:  When
forking, how do I allocate the incoming Max-Breadth to the outgoing
forks?  The obvious solution is to split it equally, but the result is
that the depth of the tree is limited to log(Max-Breadth) forks.  One
can easily construct somewhat realistic examples that demand a large
Max-Breadth in order to succeed:


                UAC  Max-Breadth = 16
                / \
               /   \  Max-Breadth = 8
              v     v
             404   proxy
                   / \
                  /   \  Max-Breadth = 4
                 v     v
                404   proxy
                      / \
                     /   \  Max-Breadth = 2
                    v     v
                   proxy 404
                   / \
                  /   \  Max-Breadth = 1
                 v     v
                200   404

That generates only 5 forks but requires a Max-Breadth of 16 to
succeed.  And under Option 2, even using serial forking has this
problem, though the message load is rather light.

However, if we choose Option 1, so that the Max-Breadth allocated to a
canceled fork can be reclaimed for use on other forks, then a proxy
can always fall back from parallel forking to serial forking.  This
allows the forking tree to be explored to Max-Forwards depth, although
extreme cases will explore the deeper leaves serially rather than
parallely, which is a user-visible performance impact (if the UASs do
not respond promptly) but doesn't cause call failure.

Dale


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From sip-bounces@ietf.org Tue Jul 24 11:11:37 2007
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On 7/24/07 9:36 AM, Jonathan Rosenberg wrote:
>
>>  The conditional header field can also be wildcarded using the special
>>    "*" entity-tag value.  Such a condition always evaluates to true
>>    regardless of the value of the current entity-tag for the resource.
>>
>>    Example:
>>
>>       Suppress-Notify-If-Match: *

We discussed this briefly at the microphone -- the answer can be 
paraphrased as "about half the stuff in Mohammad's notify-pause draft 
that we'll be discussing in SIPPING on Thursday."

If we move forward with the notify-pause stuff (which I think would be a 
good idea), then we need to have clarity around how these two mechanisms 
interact -- and whether it even makes sense in the etags document if 
we're going to develop a more complete way to address these use cases in 
the pause draft.

/a


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From sip-bounces@ietf.org Tue Jul 24 11:18:03 2007
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I would like to praise the current proposal, viz., providing each
change in two forms:  one as a textual difference to the relevant RFC
and the other as a prose narrative of the new behavior.  (There 
should also be discussion of what the difference is and what the
motivation is, but I'm not addressing that here.)  And both
descriptions should be *normative*.

The reason is that it makes the revision process much more robust
against error -- though it's unfortunately easy to make errors in
either diff form or in narrative form, it is difficult to make the
*same* error in both forms.  Contradictions between the two forms show
that an error has been made in the error correction.  (Yuck.)

This is similar to putting comments in code -- one of the major tools
for debugging code is to notice that the comments and the code do not
match.

(Designating one or the other as the sole normative form would make
the normative meaning of the correction unambiguous if there was a
contradiction, but it would not raise the probability that the
normative meaning was correct or was what was intended.  Similarly,
when a comment does not match code, either one might be wrong.)

Dale


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From sip-bounces@ietf.org Tue Jul 24 11:40:35 2007
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Subject: Re: [Sip] Two minor Max-Breadth comments
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These are important points - adding max-breadth makes the effects of  
responses from the endpoints and CANCELS (from the originator or from  
proxies) meaningful.

(In the original attack, the cancel could never catch up with the  
request).

For the last point - Throwing the 600 doesn't tear the whole tree  
down immediately. The 600 will be held by each node until all  
branches under it complete.
Once that happens it moves upstream. When it moves upstream, the next  
node starts canceling other branches. Those CANCELs still won't catch  
the request,
but the number of things it has to chase is bounded above by a constant.

RjS

On Jul 24, 2007, at 9:48 AM, Adam Roach wrote:

> During today's meeting, a point was raised surrounding whether the  
> max-breadth approach should limit the total number of requests or  
> merely the number of requests in flight at once. The statement is  
> that limiting only the ones in flight simultaneously ultimately  
> results in the same number of messages as would result without the  
> extension.
>
> While there has been a token mention of Timer C, I think we're  
> underestimating the mitigating effect of Timer C.
>
> As an additional point, I think it's worth pointing out that the  
> victim, if they recognize they are being subjected to this style of  
> attack, can tear the entire tree down fairly rapidly by throwing a  
> 600-class response.
>
> /a
>
>
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From sip-bounces@ietf.org Tue Jul 24 11:41:18 2007
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On 7/24/07 9:48 AM, Adam Roach wrote:
>
> As an additional point, I think it's worth pointing out that the 
> victim, if they recognize they are being subjected to this style of 
> attack, can tear the entire tree down fairly rapidly by throwing a 
> 600-class response.

To put a finer point on this: by "fairly rapidly," I mean "O(log(n))" 
where n=max-breadth at the root of the tree.

/a


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From sip-bounces@ietf.org Tue Jul 24 11:51:48 2007
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Inline.

ext Adam Roach wrote:
> On 7/24/07 9:36 AM, Jonathan Rosenberg wrote:
>>
>>>  The conditional header field can also be wildcarded using the special
>>>    "*" entity-tag value.  Such a condition always evaluates to true
>>>    regardless of the value of the current entity-tag for the resource.
>>>
>>>    Example:
>>>
>>>       Suppress-Notify-If-Match: *
> 
> We discussed this briefly at the microphone -- the answer can be
> paraphrased as "about half the stuff in Mohammad's notify-pause draft
> that we'll be discussing in SIPPING on Thursday."

As I said in the session today, this is not to pause a subscription at
all; it only affects the very next NOTIFY that the SUBSCRIBE is about to
generate. So I don't see where the overlap here is. (However, there is
considerable overlap with the notify-pause and the throttle draft, which
I used to author long ago when my hair was a lot shorter...)

The use-cases for the wildcard are as follows (also mentioned at the end
of 4.3.):

- Heartbeat, or liveliness check. Using Suppress-Notify-If-Match: *, the
watcher can check whether a resource is alive: a 204 response says it
is, a 404 says it is not.

- Meta-data retrieval. Just like a HEAD in HTTP, a SUBSCRIBE with
Suppress-Body-If-Match: * will generate a NOTIFY without the body but
with the full header, including the current SIP-ETag.

I agree that these are not cornerstone use-cases for this draft, but
they represent some low-hanging fruit that would be difficult to add as
an extension.

I think it is beneficial to make notifiers understand wildcards from the
very beginning.

(I believe it is also a trivial implementation effort. I mean it's a
single additional string comparison in addition to matching the given
etag value with the current etag value of the resource, which the
notifier in any case needs to perform.)

Cheers,
Aki


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From sip-bounces@ietf.org Tue Jul 24 12:05:14 2007
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Subject: [Sip] INFO, Mime-multipart, and Content Negotiation (oh, my!)
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These issues seem to be all related.  E-mail Mime can make do with
Content-Type because e-mail's sole function is to present information
to the user.  The only information needed is "What is this?" because
"What should I do with this?" is constant.

In SIP, we have the Content-Disposition, with the defined values
"session", "render", "alert", and "icon".  This answers the question
"What should I do with this?", which is independent of "What is
this?".

We've been quite sloppy about this issue because for most methods
"What should I do with this?" is strongly implied by the method or the
context or the Content-Type.

But with INFO, unless there is a tightly constrained context from some
non-SIP feature, there is complete decoupling between "What is this?"
and "What should I do with this?".  That is, any body part needs a
Content-Disposition that identifies what should be done with the
information (= what application the information should be delivered
to).

But we have no method for negotiating what Content-Dispositions are
supported by a given SIP UA.  So of course it doesn't work well.

Dale


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From sip-bounces@ietf.org Tue Jul 24 12:26:46 2007
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Date: Tue, 24 Jul 2007 11:23:24 -0500
From: Spencer Dawkins <spencer@mcsr-labs.org>
Subject: Re: [Sip] Notes on Idraft-burger-sip-info-00
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I spoke at the mike a minute ago about having this draft done in SIP, and 
then Rohan suggested that BLISS would be the right place to do the work. I 
would be OK with that, too, if others see it the same way.

Dean called the question as "do something IN THIS WG", which is fair 
(because that's the WG he chairs), but I would support "doing something IN 
THE RIGHT WG", if the chairs and ADs think there's a better home for the 
work.

Pointing to a MEDIACTRL draft is not the right thing to do, and pointing to 
a long-expired draft that people may or may not have agreed with at the time 
is even worse.

Spencer


>I finally got to look at draft-burger-sip-info-00. I like what I see. I did
> notice some nits, and am willing to provide a more formal review, but I 
> support adoption of this draft as a WG deliverable, even in its current 
> form.
>
> You may have noticed that the MEDIACTRL WG editors were forced to discuss 
> INFO in their specifications, because we currently have one draft 
> providing guidance on INFO that people seem to generally agree with, but 
> it expired years ago. I
>
> Thanks, Eric, for taking this on.
>
> Spencer 




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Hi Jonathan,

Thanks for the feedback. Inline.

ext Jonathan Rosenberg wrote:
> Overall I think this document is in great shape. Some comments below,
> mostly minor:
> 
>> Changes in connectivity represent another impetus for a subscriber
>>    re-subscribing.  If the subscriber's point of attachment to the
>>    Internet changes, e.g., due to dynamic address allocation, the
>>    subscriber needs to re-subscribe in order to update the dialog
>>    endpoint, which is carried in the Contact header field of the
>>    SUBSCRIBE request.
>>
>>       Another option for reducing connectivity induced subscription
>>       refreshes is to use the Globally Routable User Agent (UA) URI
>>       (GRUU) [10] as a stable endpoint contact for subscriptions.
> 
> Yes, exactly. And here it works really well. So I would remove this
> paragraph as a motivation.

Done.

>> The same problem affects fetching and polling of resource state as
>>    well.  As a benchmark, if we look at the performance of HTTP [9] in
>>    similar scenarios, it performs substantially better using conditional
>>    requests.
> 
> I'm curious if you have actual data on this, or just speculation. most
> HTTP resources are dynamic so its not clear it actually helps http.

I haven't looked at any empirical data on this. This comment is more
about the protocol itself, not the web as such.

And while most actual content may be dynamic, think of the dozens of
pictures that a typical web page has as backgrounds, delimiters, and
other graphics. These things rarely change, even if the actual content
of the page changes; a refresh GET really should not replay all those
jpegs.

I would make a simile here to a typical SIP notification generated
because of a refresh.

>> NOTIFY request.  The entity tag is unique for that particular
>>    resource and event notification.
> 
> You should clarify what it means to be unique for a resource *and* event
> notification. So this means if I fetch the resource with http it has a
> differnet entity tag?

My working copy says:

	The entity-tag MUST be unique across all versions of
	all entities for a resource.

I think we should also add "...and event package" just to be clear.

I think we should also add that the subscriber should never assume that
the entity-tags are shared between notification and, say, SIP event
publication or HTTP.

>> The two types of conditions available for a SUBSCRIBE are a Suppress-
>>    Body-If-Match and a Suppress-Notify-If-Match; each of these header
>>    fields contains the last entity-tag seen by the subscriber.  If the
>>    condition evaluates to true, the first of these conditions will
>>    instruct the notifier to suppress the body of the next notification
>>    and the second the entire notification. 
> 
> Is there a reason we need both? Could we just have
> Suppress-Body-If-Match and if the subscription state is anything besides
> active and the subscription duration anything different than the Expires
> in the request, a NOTIFY is sent without a body?

This would be a very drastic simplification, and it would not work for
polling, which was one of the original use-cases.

In fact, another way of doing this is to *always* use
Suppress-Body-If-Match: * in refreshes, according to the current draft.

I still believe the ability to suppress the entire NOTIFY is a valuable
feature, but as Rohan pointed out, it may also have backwards
compatibility issues, as it is a more substantial change to RFC3265.

>> There is also a wildcard entity-tag with
>>    a special value of "*" that always matches.
> 
> what is the use-case for this?
> 
> * I think you can actually implement this without option tags. The
> presence of the Suppress-* header fields basically plays that role.
> Supported is needed if the UA is not asking for some feature in a
> request, but merely saying, its supported.

The only reason it's there, actually, is for the subscriber to give a
hint to the notifier, that etags are supported. This is when a SUBSCRIBE
has no Suppress-*, i.e., it's the very first SUBSCRIBE.

However, I agree it adds very, very little value. I'll remove it.

>> uppress-Body-If-Match:  if this condition evaluates to true, the
>>       notifier is instructed to suppress (i.e., remove) the body of the
>>       first NOTIFY request following the SUBSCRIBE.  If false, the
>>       notifier follows the default behavior.
> 
> Its not 'first' - its any subsequet NOTIFY whose body, had this
> specification not be in use, would contain the resource with that entity
> tag.

Ahh, I see. Now I understand why Adam was seeing the overlap with
notify-pause.

It's 'first' intentionally. The mechanism is supposed to only affect the
very NOTIFY that the SUBSCRIBE is about to generate. It could be 'any
subsequent' but would there be any reason that a notifier replays
notifications if there's no change?

>> Suppress-Notify-If-Match:  if this condition evaluates to true, the
>>       notifier is instructed to suppress (i.e., block) the entire first
>>       NOTIFY request following the SUBSCRIBE, and instead send a 204 (No
>>       Notification) response.  If false, the notifier again follows the
>>       default behavior.
> 
> I don't think we need a new response code. What is the justification?

It is to enable the subscriber to destroy its handle as soon as the 204
is received, instead of keeping it indefinitely in case there is a
NOTIFY forthcoming (it may not ever come).

> * Section 4 needs to cover handling of the response to SUBSCRIBE (what
> happens when a 204 comes), and NOTIFY (store entity tag). Also needs to
> handle what to do in the case where the notifier doesn't support the
> mechanism.

I can add text to the numbered lists of 4.x sections about storing the
entity tags. The very last bullets of each of 4.x sections do talk about
what the subscriber is to do with the 204. Should there be a sub-section
in section 4 that has normative text on these though?

> 
>> A notifier MUST generate entity tags for each resource it is
>>    responsible for.
>>
>>       The views might correspond to different groups of users that have
>>       varying levels of access rights to the resource state, or to
>>       subscribers that have modified their subscription using event
>>       notification filtering [11].  For example, in presence [6]
>>       watchers may get different levels of accuracy in geolocation
>>       information, based on the presentity's privacy settings.
> 
> What do views have to do with this?

Here view == entity, but clearly the view should be in double quotes, as
it's not a term that is clearly defined anywhere. However, The next
version will have section 5.1. strengthened to make this part of the
model (hopefully) clear.

> Nits:
> 
> 
> 
>> These procedures have a serious
>>    deficiency in that they do not allow state to persist over a
>>    subscription refresh, or between two consecutive polls. 
> 
> State can persist at the UA; its just that you get notified of stuff you
> already know.

True. I'll reword:

...deficiency in that they provide no tools to avoid replaying event
notifications that are already known to the UA.

>>  An entity-tag is a token carried in the SIP-ETag header field, and it
>>    is opaque to the client.  The notifier is free to decide the means
>>    for generating an entity-tag, except for the special "*" value.
> 
> I think you mean, "The notifier is free to decide on any means for
> generating the entity-tag. It can have any value, except for "*""

True, this is already fixed in my working copy. Dale made the same
correction.

Cheers,
Aki


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I was thinking the same thing. Dealing with this in the context of 
"features" that use it has some promise of at least demonstrating the 
problems with existing approaches to those that matter.

	Paul

Spencer Dawkins wrote:
> I spoke at the mike a minute ago about having this draft done in SIP, 
> and then Rohan suggested that BLISS would be the right place to do the 
> work. I would be OK with that, too, if others see it the same way.
> 
> Dean called the question as "do something IN THIS WG", which is fair 
> (because that's the WG he chairs), but I would support "doing something 
> IN THE RIGHT WG", if the chairs and ADs think there's a better home for 
> the work.
> 
> Pointing to a MEDIACTRL draft is not the right thing to do, and pointing 
> to a long-expired draft that people may or may not have agreed with at 
> the time is even worse.
> 
> Spencer
> 
> 
>> I finally got to look at draft-burger-sip-info-00. I like what I see. 
>> I did
>> notice some nits, and am willing to provide a more formal review, but 
>> I support adoption of this draft as a WG deliverable, even in its 
>> current form.
>>
>> You may have noticed that the MEDIACTRL WG editors were forced to 
>> discuss INFO in their specifications, because we currently have one 
>> draft providing guidance on INFO that people seem to generally agree 
>> with, but it expired years ago. I
>>
>> Thanks, Eric, for taking this on.
>>
>> Spencer 
> 
> 
> 
> 
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> 


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Subject: RE: [Sip] INFO and Content Negotiation
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Inline...

> -----Original Message-----
> From: Eric Burger [mailto:eburger@bea.com]
> Sent: Tuesday, July 24, 2007 8:16 AM
> To: Xavier Marjou
> Cc: IETF SIP
> Subject: Re: [Sip] INFO and Content Negotiation
> 
> Indicating the package is possible; it is what Dean's draft of 4 years ago
> proposed.  However, it is not what is there today.  Moreover, it is
> doubtful
> that any of the myriad INFO exploits would use the mechanism.  If one is
> going to upgrade one's mechanism, one might as well do the right thing.

I'm actually not convinced of that.  I don't think people only use INFO
because it was convenient for experimentation, and that given the choice of
modifying it to fix some issues vs. completely changing how they implement
feature XYZ, that they would choose the latter.  I think they use INFO
because it achieves what they need to do in the simplest and most scalable
way, and the KISS principle is strong motivation for many.  

In particular, for the DTMF use of INFO my guess is many vendors would be
willing to add explicit negotiation, rather than doing subscribe-KPML. (so
long as the negotiation itself was simple)

-hadriel




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Adam,

=20

I fully agree with your positive assessment of the ICE-17 document.

=20

Do you by any chance plan to also develop a test tool for testing the
NAT environment and the success/failure cases, connection times, etc.,
to measure results?

=20

Or maybe you know someone who does?

=20

Thanks, Henry

=20

________________________________

From: adamfisk@gmail.com [mailto:adamfisk@gmail.com] On Behalf Of Adam
Fisk
Sent: Tuesday, July 24, 2007 3:08 PM
To: Jiri Kuthan
Cc: Henry Sinnreich; David Barrett; Jan Janak; Juha Heinanen;
sip@ietf.org; mmusic@ietf.org
Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
deploymentdatabefore LC for RFC

=20

I've followed ICE development on the sidelines for years now, and I'm
currently implementing ICE 17.  I've implemented what will be open
source implementations of STUN and TURN, both client and server.  Here
are the key points I've taken away from that experience:=20

1) The patience of Jonathan and everyone else in meticulously getting to
the current draft is astonishing.
2) ICE is complex as hell and a pain in the ^#* to implement.  I was
surprised at how hard it was, particularly for a draft that's so close
to becoming an RFC.=20
3) ICE is solving a very tricky problem.

I like Phillip's earlier suggestion that the draft stay in the "almost
done" state like used to happen in the Routing group.  I realize that's
asking a lot given how much work has gone into it and how eager everyone
is to finalize it.  I think the problem is that standardizing NAT
traversal is just hard, and hard problems are difficult to make simpler.
In my implementation of ICE, I do think it could be done more simply,
but I don't have concrete suggestions right now because I'm still
pulling my hair out with the implementation.  I'm not talking about
drastic changes anywhere, but all the checklist fiddling sure takes a
lot of patience to write.  Simplifications might just be in the
description and not necessarily the algorithm.=20

I think it's a hard enough and important enough problem that we owe it
to ourselves to be even more patient with an "almost done" state that
adds the final touches based on real-world feedback.  I don't doubt for
a second that ICE works, but it's so complex that I worry no one outside
huge companies will ever implement it.  That would be a huge hindrance
to ICE's adoption and power.=20

-Adam


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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'>Adam,<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 fully agree with your positive =
assessment
of the ICE-17 document.<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'>&nbsp;<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'>Do you by any chance plan to also =
develop
a test tool for testing the NAT environment and the success/failure =
cases, connection
times, etc., to measure results?<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'>Or maybe you know someone who =
does?<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, =
Henry<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'>
adamfisk@gmail.com [mailto:adamfisk@gmail.com] <b><span =
style=3D'font-weight:
bold'>On Behalf Of </span></b>Adam Fisk<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, July 24, =
2007 3:08
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Jiri Kuthan<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <st1:PersonName =
w:st=3D"on">Henry
 Sinnreich</st1:PersonName>; <st1:PersonName w:st=3D"on">David =
Barrett</st1:PersonName>;
<st1:PersonName w:st=3D"on">Jan Janak</st1:PersonName>; <st1:PersonName =
w:st=3D"on">Juha
 Heinanen</st1:PersonName>; <st1:PersonName =
w:st=3D"on">sip@ietf.o<st1:PersonName
 w:st=3D"on">rg</st1:PersonName></st1:PersonName>; <st1:PersonName =
w:st=3D"on">mmusic@ietf.o<st1:PersonName
 w:st=3D"on">rg</st1:PersonName></st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [MMUSIC] Re: =
[Sip]
RE: [BEHAVE] Re: ICE deploymentdatabefore LC for =
RFC</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 style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>I've followed =
ICE
development on the sidelines for years now, and I'm currently =
implementing ICE
17.&nbsp; I've implemented what will be open source implementations of =
STUN and
TURN, both client and server.&nbsp; Here are the key points I've taken =
away
from that experience: <br>
<br>
1) The patience of Jonathan and everyone else in meticulously getting to =
the
current draft is astonishing.<br>
2) ICE is complex as hell and a pain in the ^#* to implement.&nbsp; I =
was
surprised at how hard it was, particularly for a draft that's so close =
to
becoming an RFC. <br>
3) ICE is solving a very tricky problem.<br>
<br>
I like Phillip's earlier suggestion that the draft stay in the =
&quot;almost
done&quot; state like used to happen in the Routing group.&nbsp; I =
realize
that's asking a lot given how much work has gone into it and how eager =
everyone
is to finalize it.&nbsp; I think the problem is that standardizing NAT =
traversal
is just hard, and hard problems are difficult to make simpler.&nbsp; In =
my
implementation of ICE, I do think it could be done more simply, but I =
don't
have concrete suggestions right now because I'm still pulling my hair =
out with
the implementation.&nbsp; I'm not talking about drastic changes =
anywhere, but
all the checklist fiddling sure takes a lot of patience to write.&nbsp;
Simplifications might just be in the description and not necessarily the
algorithm. <br>
<br>
I think it's a hard enough and important enough problem that we owe it =
to
ourselves to be even more patient with an &quot;almost done&quot; state =
that
adds the final touches based on real-world feedback.&nbsp; I don't doubt =
for a
second that ICE works, but it's so complex that I worry no one outside =
huge
companies will ever implement it.&nbsp; That would be a huge hindrance =
to ICE's
adoption and power. <br>
<br>
-Adam<o:p></o:p></span></font></p>

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Date: Tue, 24 Jul 2007 16:07:32 -0400
From: "Adam Fisk" <a@lastbamboo.org>
To: "Jiri Kuthan" <jiri@iptel.org>
Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deploymentdatabefore LC
	for RFC
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I've followed ICE development on the sidelines for years now, and I'm
currently implementing ICE 17.  I've implemented what will be open source
implementations of STUN and TURN, both client and server.  Here are the key
points I've taken away from that experience:

1) The patience of Jonathan and everyone else in meticulously getting to the
current draft is astonishing.
2) ICE is complex as hell and a pain in the ^#* to implement.  I was
surprised at how hard it was, particularly for a draft that's so close to
becoming an RFC.
3) ICE is solving a very tricky problem.

I like Phillip's earlier suggestion that the draft stay in the "almost done"
state like used to happen in the Routing group.  I realize that's asking a
lot given how much work has gone into it and how eager everyone is to
finalize it.  I think the problem is that standardizing NAT traversal is
just hard, and hard problems are difficult to make simpler.  In my
implementation of ICE, I do think it could be done more simply, but I don't
have concrete suggestions right now because I'm still pulling my hair out
with the implementation.  I'm not talking about drastic changes anywhere,
but all the checklist fiddling sure takes a lot of patience to write.
Simplifications might just be in the description and not necessarily the
algorithm.

I think it's a hard enough and important enough problem that we owe it to
ourselves to be even more patient with an "almost done" state that adds the
final touches based on real-world feedback.  I don't doubt for a second that
ICE works, but it's so complex that I worry no one outside huge companies
will ever implement it.  That would be a huge hindrance to ICE's adoption
and power.

-Adam

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I&#39;ve followed ICE development on the sidelines for years now, and I&#39;m currently implementing ICE 17.&nbsp; I&#39;ve implemented what will be open source implementations of STUN and TURN, both client and server.&nbsp; Here are the key points I&#39;ve taken away from that experience:
<br><br>1) The patience of Jonathan and everyone else in meticulously getting to the current draft is astonishing.<br>2) ICE is complex as hell and a pain in the ^#* to implement.&nbsp; I was surprised at how hard it was, particularly for a draft that&#39;s so close to becoming an RFC.
<br>3) ICE is solving a very tricky problem.<br><br>I like Phillip&#39;s earlier suggestion that the draft stay in the &quot;almost done&quot; state like used to happen in the Routing group.&nbsp; I realize that&#39;s asking a lot given how much work has gone into it and how eager everyone is to finalize it.&nbsp; I think the problem is that standardizing NAT traversal is just hard, and hard problems are difficult to make simpler.&nbsp; In my implementation of ICE, I do think it could be done more simply, but I don&#39;t have concrete suggestions right now because I&#39;m still pulling my hair out with the implementation.&nbsp; I&#39;m not talking about drastic changes anywhere, but all the checklist fiddling sure takes a lot of patience to write.&nbsp; Simplifications might just be in the description and not necessarily the algorithm.
<br><br>I think it&#39;s a hard enough and important enough problem that we owe it to ourselves to be even more patient with an &quot;almost done&quot; state that adds the final touches based on real-world feedback.&nbsp; I don&#39;t doubt for a second that ICE works, but it&#39;s so complex that I worry no one outside huge companies will ever implement it.&nbsp; That would be a huge hindrance to ICE&#39;s adoption and power.
<br><br>-Adam<br><br><br>

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From sip-bounces@ietf.org Tue Jul 24 17:15:26 2007
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Subject: Re: [Sip] INFO, Mime-multipart, and Content Negotiation (oh, my!)
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   From: Dale.Worley@comcast.net

   These issues seem to be all related.  E-mail Mime can make do with
   Content-Type because e-mail's sole function is to present information
   to the user.  The only information needed is "What is this?" because
   "What should I do with this?" is constant.

   In SIP, we have the Content-Disposition, with the defined values
   "session", "render", "alert", and "icon".  This answers the question
   "What should I do with this?", which is independent of "What is
   this?".

   We've been quite sloppy about this issue because for most methods
   "What should I do with this?" is strongly implied by the method or the
   context or the Content-Type.

   But with INFO, unless there is a tightly constrained context from some
   non-SIP feature, there is complete decoupling between "What is this?"
   and "What should I do with this?".  That is, any body part needs a
   Content-Disposition that identifies what should be done with the
   information (= what application the information should be delivered
   to).

   But we have no method for negotiating what Content-Dispositions are
   supported by a given SIP UA.  So of course it doesn't work well.

   Dale


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I sent that last message too quickly:

   From: Dale.Worley@comcast.net

   But we have no method for negotiating what Content-Dispositions are
   supported by a given SIP UA.  So of course it doesn't work well.

But as someone suggested at the microphone, if people coin new methods
(starting with "X-"), those methods *can* encode "What should I do
with this?"  And the Allow header allows UAs to negotiate what methods
are supported.

I see no problem with proliferating methods, as that is no worse than
proliferating Content-Dispositions.

Dale


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From sip-bounces@ietf.org Tue Jul 24 17:25:13 2007
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At 22:07 24/07/2007, Adam Fisk wrote:
>I've followed ICE development on the sidelines for years now, and I'm currently implementing ICE 17.  I've implemented what will be open source implementations of STUN and TURN, both client and server.

That's marvellous news -- can you send me a pointer to that?

>  Here are the key points I've taken away from that experience: 
>
>1) The patience of Jonathan and everyone else in meticulously getting to the current draft is astonishing.
>2) ICE is complex as hell and a pain in the ^#* to implement.  I was surprised at how hard it was, particularly for a draft that's so close to becoming an RFC. 
>3) ICE is solving a very tricky problem.

All three points are wholeheartly easy to agree with. I would have added too
ICE is the "right" solution.

The question to me is whether Mr. Time and Mr. $ are not against us. That question
was asked couple of meetings ago by Spencer D. in an 'sbc bar bof' -- basically 
fingerpointing at the NAT experience, "right" solution (IPv6) and the outcome. 
The concern is that by time we have ICE, everyone will have the problem solved 
using media relays and mass economy will make it a more viable solution than anything 
else.

Future is generally hard to predict (that's why analysts like to predict past :-)),
so I think it would be at least a proper step to document current practices (media
relays) so that the IETF's picture stays grounded (whatever the risky comeout of work 
on "right" solutions may be.)

-jiri



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From sip-bounces@ietf.org Tue Jul 24 17:28:53 2007
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Subject: Re: [Sip] INFO, Mime-multipart, and Content Negotiation (oh, my!)
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> I see no problem with proliferating methods, as that is no worse than
> proliferating Content-Dispositions.
>

Well, for one it could cause existing proxies to Record-Route such requests 
as they would not know the method and would therefore be forced to assume it 
to be dialog creating. So there is some overhead, compared to C-D as 
extension point

Also, in practice I would expect unknown content-dispositions to interwork 
better than unknown methods, especially through proxies

Regards,
Jeroen 



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From sip-bounces@ietf.org Tue Jul 24 20:21:40 2007
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To: jdrosen@cisco.com
Subject: Re: [Sip] comments on draft-munakata-sip-privacy-new
From: Mayumi Munakata <munakata.mayumi@lab.ntt.co.jp>
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Jonathan,

Thank you for the comments for the privacy-new draft.
See inline.

> Section 1: There are some additional problems worth noting:
> * privacy service has to be a b2bua and thus interferes with security mechanisms that break in the face of b2bua, such as sip identity (which provides message integrity and works well here too to provide the remote side an assurance of authenticated identity, just one that is not provided), which in turn breaks dtls-srtp
>
> * because of the need to signal from the UA to the privacy service exactly the desired functionality, the set of functionalities possible, in terms of what is anonyzmized and what is not, is bounded and required detailed specification.

Will add this in the section 1.

>
>
>> The user agent setting the "temp-gruu" as a GRUU SHOULD set
>> "Anonymous" as a display name in any header where the display name of
>> the originator is set. That indicates the anonymity of the request
>> to intermediaries that may invoke some services based on the
>> anonymity of the call.
>
> Hmm, I'd prefer something explicit. There are problems with using human-readable and locale-specific text like display name to signal to automata.

"Anonymous" display name is not used for explicit indication of anonymity to the
intermediaries, rather this is specified to be aligned with the ACR draft. AFAIK
the ACR draft explicitly states the spelling of display-name to be either "Anonymous"
or "anonymous".

>
>> Note: The mechanism of the flag is FFS.
>
> From text above I thought it was presence of the Anonymous in display name, which is bad. I personally like the Privacy: id value since its effectively what everyone is using and also backwards compatible.

This is based on few assumption.
1. privacy value beside "id" is not used.
>> This sounds almost true, but I have heard several vendors and providers actually utilizing these values to
micro manage the privacy treatment of messages, in which case overloading the semantics of "id" may
be problematic when older privacy service is treating the message as noted on the slide today.

2. information beside what's inserted by the UA is not a privacy concern.
>> Information readily added by the intermediaries such as Record-Route, Via etc. are not privacy
sensitive. You have to be making this assumption to overload the "id", as it is possible that older
privacy service would only remove the P-A-ID and forward the listed values above which are
considered to be a target of privacy in RFC 3323.

Anyhow, without thorough knowledge of what's really deployed in the wild, it is rather dangerous to
make any assumption and overload the "id" value.

>
> I also think we need something to make sure the terminating UA doesn't try to render the From or P-A-ID if someone should insert.

Good point.

AFAIK, the Privacy header is removed when the last value for the privacy header is removed so something is needed. Having that said, you seemed to be implying that something needs to be there to convey that UAC is requesting the privacy, yet survives translation and treatment of privacy service and other intermediaries.

>
>> Proxy Behavior
>
> The draft should talk about RFC 3325 in particular.

Okay.

>
> Nits:
>
>
>> privacy service. This document proposes a new privacy mechanism that
>> a user agent can facilitate to conceal privacy-sensitive information
>> without the need for aid from a privacy service.
>
> This is somehwat misleading; there is a privacy service in the form of turn servers and proxies.

Right, I think we simply copied the terminology from the RFC3323, will modify the text to
be more concise.

>
>> The protection of network
>> privacy (e.g., topology hiding) is outside the scope of this
>> document.
>
> You need to clarify what this means.

Will do.

>
>> A user agent wanting to obtain functional anonymous URI SHOULD
>> support and SHOULD utilize the Global Routable User Agent URI (GRUU)
>> mechanism. 
>
> Isn't this going to need to be a MUST?

I guess we will rewrite it to say, URI SHOULD support and if support
MUST utilize the GRUU mechanism.

>
>> response for a "temp-gruu" URI parameter, which provides the desired
>> privacy property.
> provides URI with the desired property

Will fix.

>
>> It is assumed that a user agent is either manually or automatically
>> configured through means such as a configuration framework with one
>> or more STUN relay servers.
>
> terminology has changed again; these are now called TURN servers once more.

Will fix.

Thanks,
Mayumi

>
>
>
> Thanks,
> Jonathan R.
>



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   From: "Jeroen van Bemmel" <jbemmel@zonnet.nl>

   > I see no problem with proliferating methods, as that is no worse than
   > proliferating Content-Dispositions.

   Well, for one it could cause existing proxies to Record-Route such
   requests as they would not know the method and would therefore be
   forced to assume it to be dialog creating. So there is some
   overhead, compared to C-D as extension point

IIRC, there's a set of rules that avoid Record-Routing except in the
first request or two of a dialog.  Scott Lawrence once explained them
to me, and I checked them, and they're method-independent (as one
would expect).

   Also, in practice I would expect unknown content-dispositions to interwork 
   better than unknown methods, especially through proxies

Proxies shouldn't care about the method.  SBCs might be more trouble,
but I expect quality SBCs to be method-agnostic too.

Dale



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Subject: [Sip] Doubt about ACK and BYE 
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Hi SIP-experts,
   
  I have doubt in a specific Scenario where SIP client is set to send INVITE messages via outbound proxy. The setup is like below:
   
   
  SIP-client 1---------------------------SIP outbound---------------------SIP proxy2---------------SIP client2
                                                  proxy1
   
  Here SIP client1 is registered with Proxy2 and set outbound proxy as proxy1. Outbound proxy1 is set not to include its ip address in the Record-Route header. After SIP 200 OK message is received in SIP-client1 which is still sending ACK and BYE messages via outbound proxy1 instead proxy2.
   
  Could you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?
   
  Regards
  Sengottuvelan
   
   
   

       
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<div>Hi SIP-experts,</div>  <div>&nbsp;</div>  <div>I have doubt in a specific <SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Scenario where SIP client is set to send INVITE messages via outbound proxy. The setup is like below:</SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">SIP-client 1---------------------------SIP outbound---------------------SIP proxy2---------------SIP client2</SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; 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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;proxy1</SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY:
 Arial"><?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Here SIP client1 is registered&nbsp;with Proxy2 and set outbound proxy as proxy1. Outbound proxy1 is set not to include its ip address in the Record-Route header. After SIP 200 OK message is received in&nbsp;SIP-client1 which&nbsp;is still sending ACK and BYE messages via outbound proxy1 instead proxy2.</o:p></SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Could you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?</o:p></SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Regards</o:p></SPAN></div>  <div><SPAN style="FONT-SIZE:
 10pt; FONT-FAMILY: Arial"><o:p>Sengottuvelan</o:p></SPAN></div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN>&nbsp;</div>  <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN>&nbsp;</div><p>&#32;
      <hr size=1>Shape Yahoo! in your own image.  
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Subject: RE: [Sip] RE: Correction: WGLC for draft-ietf-sip-outbound
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Robert Sparks" <rjsparks@nostrum.com>
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Just to indicate that what I don't think we want during the middle of
the WGLC is a -11 version. I think we need all comments made during this
last call to be made against -10.

What I would like is a compilation of all the comments made like we did
for GRUU - they are still up on softarmor. I would consider that we
could start plugging the proposed resolution into such a document.

I don't know what Dean's availability is in this respect, but I haven't
got time to devote to this until late next week.=20

Regards

Keith

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Tuesday, July 24, 2007 2:17 PM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org; Cullen Jennings; Rohan Mahy
> Subject: Re: [Sip] RE: Correction: WGLC for draft-ietf-sip-outbound
>=20
> Keith/Rohan -
>=20
> There's a moderate list of changes we already know are going=20
> to be made to the document.
>=20
> Everyone who wasn't in Chicago's Monday SIP meeting should=20
> look at=20
> http://www3.ietf.org/proceedings/07jul/slides/sip-3.pdf for=20
> the bigger ones.
> Rohan has indicated a few smaller ones on list.
>=20
> Would it be possible to get a list of the known expected=20
> changes in one place?
>=20
> RjS
>=20
> On Jul 23, 2007, at 11:23 PM, DRAGE, Keith ((Keith)) wrote:
>=20
> > A reminder of the ongoing WGLC on the above document
> >
> > This WGLC period runs through IETF and hopefully close=20
> August 6, 2007.
> >
> > That gives you about a week after the end of IETF - please schedule=20
> > some time for doing this.
> >
> > Regards
> >
> > Keith
> >
> >
> >> -----Original Message-----
> >> From: Dean Willis [mailto:dean.willis@softarmor.com]
> >> Sent: Thursday, July 12, 2007 7:14 PM
> >> To: Dean Willis
> >> Cc: sip@ietf.org; Cullen Jennings; DRAGE, Keith (Keith)
> >> Subject: Correction: WGLC for draft-ietf-sip-outbound
> >>
> >> Dean Willis wrote:
> >>> I'd like to start a long working group last call for:
> >>>
> >>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
> >>>
> >>
> >> My error. This should have referred to draft-ietf-sip-outbound-10,=20
> >> not -09. It was busy working its way through the posting=20
> system and I=20
> >> grabbed a pointer to the wrong one.
> >>
> >> Sorry about that, folks.
> >>
> >> --
> >> Dean
> >>
> >>
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20


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Subject: RE: [Sip] Two minor Max-Breadth comments
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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How does the UAC cancelling the request help.

Surely all we end up with is the CANCEL request desperatly trying to
catch up with all the propagated INVITE requests. I see nothing in the
transaction model that helps it catch up, and I suspect (based on
experience in non-SIP systems) that it never will in most people's
implementations. It will always be pretty much the same number of steps
behind.

Regards

Keith=20

> -----Original Message-----
> From: Dale Worley [mailto:worley@dworley.hsd1.ma.comcast.net]=20
> Sent: Tuesday, July 24, 2007 4:01 PM
> To: sip@ietf.org
> Subject: Re: [Sip] Two minor Max-Breadth comments
>=20
>    From: Adam Roach <adam@nostrum.com>
>=20
>    During today's meeting, a point was raised surrounding whether the=20
>    max-breadth approach should limit the total number of=20
> requests or merely=20
>    the number of requests in flight at once. The statement is=20
> that limiting=20
>    only the ones in flight simultaneously ultimately results=20
> in the same=20
>    number of messages as would result without the extension.
>=20
>    While there has been a token mention of Timer C, I think we're=20
>    underestimating the mitigating effect of Timer C.
>=20
>    As an additional point, I think it's worth pointing out=20
> that the victim,=20
>    if they recognize they are being subjected to this style=20
> of attack, can=20
>    tear the entire tree down fairly rapidly by throwing a=20
> 600-class response.
>=20
> Also, the UAC, if it is not perverse, will ultimately cancel=20
> the request.
>=20
> Dale
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Just to indicate that what I don't think we want during the middle of
the WGLC is a -11 version. I think we need all comments made during this
last call to be made against -10.

What I would like is a compilation of all the comments made like we did
for GRUU - they are still up on softarmor. I would consider that we
could start plugging the proposed resolution into such a document.

I don't know what Dean's availability is in this respect, but I haven't
got time to devote to this until late next week.=20

Regards

Keith

> -----Original Message-----
> From: Robert Sparks [mailto:rjsparks@nostrum.com]=20
> Sent: Tuesday, July 24, 2007 2:17 PM
> To: DRAGE, Keith (Keith)
> Cc: sip@ietf.org; Cullen Jennings; Rohan Mahy
> Subject: Re: [Sip] RE: Correction: WGLC for draft-ietf-sip-outbound
>=20
> Keith/Rohan -
>=20
> There's a moderate list of changes we already know are going=20
> to be made to the document.
>=20
> Everyone who wasn't in Chicago's Monday SIP meeting should=20
> look at=20
> http://www3.ietf.org/proceedings/07jul/slides/sip-3.pdf for=20
> the bigger ones.
> Rohan has indicated a few smaller ones on list.
>=20
> Would it be possible to get a list of the known expected=20
> changes in one place?
>=20
> RjS
>=20
> On Jul 23, 2007, at 11:23 PM, DRAGE, Keith ((Keith)) wrote:
>=20
> > A reminder of the ongoing WGLC on the above document
> >
> > This WGLC period runs through IETF and hopefully close=20
> August 6, 2007.
> >
> > That gives you about a week after the end of IETF - please schedule=20
> > some time for doing this.
> >
> > Regards
> >
> > Keith
> >
> >
> >> -----Original Message-----
> >> From: Dean Willis [mailto:dean.willis@softarmor.com]
> >> Sent: Thursday, July 12, 2007 7:14 PM
> >> To: Dean Willis
> >> Cc: sip@ietf.org; Cullen Jennings; DRAGE, Keith (Keith)
> >> Subject: Correction: WGLC for draft-ietf-sip-outbound
> >>
> >> Dean Willis wrote:
> >>> I'd like to start a long working group last call for:
> >>>
> >>> http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-09.txt
> >>>
> >>
> >> My error. This should have referred to draft-ietf-sip-outbound-10,=20
> >> not -09. It was busy working its way through the posting=20
> system and I=20
> >> grabbed a pointer to the wrong one.
> >>
> >> Sorry about that, folks.
> >>
> >> --
> >> Dean
> >>
> >>
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Wed Jul 25 03:08:34 2007
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Subject: RE: [Sip] Two minor Max-Breadth comments
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Dale Worley" <worley@dworley.hsd1.ma.comcast.net>, <sip@ietf.org>
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How does the UAC cancelling the request help.

Surely all we end up with is the CANCEL request desperatly trying to
catch up with all the propagated INVITE requests. I see nothing in the
transaction model that helps it catch up, and I suspect (based on
experience in non-SIP systems) that it never will in most people's
implementations. It will always be pretty much the same number of steps
behind.

Regards

Keith=20

> -----Original Message-----
> From: Dale Worley [mailto:worley@dworley.hsd1.ma.comcast.net]=20
> Sent: Tuesday, July 24, 2007 4:01 PM
> To: sip@ietf.org
> Subject: Re: [Sip] Two minor Max-Breadth comments
>=20
>    From: Adam Roach <adam@nostrum.com>
>=20
>    During today's meeting, a point was raised surrounding whether the=20
>    max-breadth approach should limit the total number of=20
> requests or merely=20
>    the number of requests in flight at once. The statement is=20
> that limiting=20
>    only the ones in flight simultaneously ultimately results=20
> in the same=20
>    number of messages as would result without the extension.
>=20
>    While there has been a token mention of Timer C, I think we're=20
>    underestimating the mitigating effect of Timer C.
>=20
>    As an additional point, I think it's worth pointing out=20
> that the victim,=20
>    if they recognize they are being subjected to this style=20
> of attack, can=20
>    tear the entire tree down fairly rapidly by throwing a=20
> 600-class response.
>=20
> Also, the UAC, if it is not perverse, will ultimately cancel=20
> the request.
>=20
> Dale
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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=20
It is some configuration issue (or maybe a bug) in the SIP client 1 I
think.
=20
Once the call is established and the route has been learnt (using
Record-Route), any new
requests should only be sent to the proxies on that learnt route.
Unfortunately I have heard that some SIP clients break this rule and
send EVERY message to
the outbound proxy even when the outbound proxy did not add a
Record-Route header.
=20
>>Could you anyone clarify that how can i send ACK and BYE messages to
Registered proxy2 instead outbound proxy1?
Of course, it should only get sent to proxy 2 if proxy 2 had added a
Record-Route.
Like I say, to fix the problem, I think it's either a bug or
configuration issue for SIP client 1.
=20
Regards,

Attila
=20
=20


________________________________

From: sengottuvelan srirangan [mailto:sengottuvelan_s@yahoo.com]=20
Sent: 25 July 2007 06:15
To: sip@ietf.org
Subject: [Sip] Doubt about ACK and BYE=20


Hi SIP-experts,
=20
I have doubt in a specific Scenario where SIP client is set to send
INVITE messages via outbound proxy. The setup is like below:
SIP-client 1---------------------------SIP
outbound---------------------SIP proxy2---------------SIP client2
                                           &n! bsp;    proxy1
=20
Here SIP client1 is registered with Proxy2 and set outbound proxy as
proxy1. Outbound proxy1 is set not to include its ip address in the
Record-Route header. After SIP 200 OK message is received in SIP-client1
which is still sending ACK and BYE messages via outbound proxy1 instead
proxy2.
=20
Could you anyone clarify that how can i send ACK and BYE messages to
Registered proxy2 instead outbound proxy1?
=20
Regards
Sengottuvelan
=20
=20
=20

________________________________

Shape Yahoo! in your own image. Join our Network Research Panel today!
<http://us.rd.yahoo.com/evt=3D48517/*http://surveylink.yahoo.com/gmrs/yah=
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML xmlns:o =3D "urn:schemas-microsoft-com:office:office"><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2873" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>It is some configuration issue (or maybe a bug) =
in the SIP=20
client 1 I think.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Once the call is established and the route has =
been learnt=20
(using Record-Route), any new</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>requests should only be sent to the proxies on =
that=20
learnt&nbsp;route.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Unfortunately I have heard that some SIP =
clients break this=20
rule and send&nbsp;EVERY message to</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>the outbound proxy </FONT></SPAN><SPAN=20
class=3D220540807-25072007><FONT face=3DArial color=3D#0000ff =
size=3D2>even when=20
the&nbsp;outbound proxy did not add a Record-Route =
header.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D220540807-25072007>&gt;&gt;Could you anyone=20
clarify that how can i send ACK and BYE messages to Registered proxy2 =
instead=20
outbound proxy1?</DEFANGHTML_SPAN></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Of course, it should only get sent to proxy 2 =
if proxy 2=20
had added a Record-Route.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Like I say, to fix the problem, I think it's =
either a bug=20
or configuration issue for SIP client 1.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV><SPAN=20
class=3D220540807-25072007><FONT face=3DArial color=3D#0000ff size=3D2>
<DIV dir=3Dltr align=3Dleft><BR>Attila</DIV>
<DIV dir=3Dltr align=3Dleft></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D220540807-25072007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sengottuvelan srirangan=20
[mailto:sengottuvelan_s@yahoo.com] <BR><B>Sent:</B> 25 July 2007=20
06:15<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Doubt about =
ACK and=20
BYE <BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Hi SIP-experts,</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV>I have doubt in a specific <DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Scenario where SIP client =
is set to=20
send INVITE messages via outbound proxy. The setup is like=20
below:</DEFANGHTML_SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><DEFANGHTML_SPAN =

style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></DEFANGHTML_SPAN></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><DEFANGHTML_SPAN =

style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">SIP-client=20
1---------------------------SIP outbound---------------------SIP=20
proxy2---------------SIP client2</DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; 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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&amp;n!=20
bsp;&nbsp;&nbsp;&nbsp;&nbsp;proxy1</DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>
<DIV><DEFANGHTML_SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>Here SIP=20
client1 is registered&nbsp;with Proxy2 and set outbound proxy as proxy1. =

Outbound proxy1 is set not to include its ip address in the Record-Route =
header.=20
After SIP 200 OK message is received in&nbsp;SIP-client1 which&nbsp;is =
still=20
sending ACK and BYE messages via outbound proxy1 instead=20
proxy2.</o:p></DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>
<DIV><DEFANGHTML_SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>Could you=20
anyone clarify that how can i send ACK and BYE messages to Registered =
proxy2=20
instead outbound proxy1?</o:p></DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>
<DIV><DEFANGHTML_SPAN tyle=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"=20
s!><o:p>Regards</o:p></DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>Sengottuvelan</o:p></DEFANGHTML_SPAN></DIV>
<DIV><DEFANGHTML_SPAN=20
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From: sengottuvelan srirangan <sengottuvelan_s@yahoo.com>
Subject: RE: [Sip] Doubt about ACK and BYE 
To: Attila Sipos <attila.sipos@vegastream.com>, sip@ietf.org
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Hi Attila,
   
  The problem is got fixed after changing the client1(as you suggested).
   
  Thanks.
   
  Regards
  Sengottuvelan

Attila Sipos <attila.sipos@vegastream.com> wrote:
       
  It is some configuration issue (or maybe a bug) in the SIP client 1 I think.
   
  Once the call is established and the route has been learnt (using Record-Route), any new
  requests should only be sent to the proxies on that learnt route.
  Unfortunately I have heard that some SIP clients break this rule and send EVERY message to
  the outbound proxy even when the outbound proxy did not add a Record-Route header.
   
  >>Could you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?
  Of course, it should only get sent to proxy 2 if proxy 2 had added a Record-Route.
  Like I say, to fix the problem, I think it's either a bug or configuration issue for SIP client 1.
   
  Regards,
  
Attila
   
   

    
---------------------------------
  From: sengottuvelan srirangan [mailto:sengottuvelan_s@yahoo.com] 
Sent: 25 July 2007 06:15
To: sip@ietf.org
Subject: [Sip] Doubt about ACK and BYE 


  
  Hi SIP-experts,
   
  I have doubt in a specific Scenario where SIP client is set to send INVITE messages via outbound proxy. The setup is like below:
  
  
  SIP-client 1---------------------------SIP outbound---------------------SIP proxy2---------------SIP client2
                                             &n! bsp;    proxy1
   
  Here SIP client1 is registered with Proxy2 and set outbound proxy as proxy1. Outbound proxy1 is set not to include its ip address in the Record-Route header. After SIP 200 OK message is received in SIP-client1 which is still sending ACK and BYE messages via outbound proxy1 instead proxy2.
   
  Could you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?
   
  Regards
  Sengottuvelan
   
   
   
    
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<div>Hi Attila,</div>  <div>&nbsp;</div>  <div>The problem is got fixed after changing the client1(as you suggested).</div>  <div>&nbsp;</div>  <div>Thanks.</div>  <div>&nbsp;</div>  <div>Regards</div>  <div>Sengottuvelan<BR><BR><B><I>Attila Sipos &lt;attila.sipos@vegastream.com&gt;</I></B> wrote:</div>  <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">  <META content="MSHTML 6.00.2900.2873" name=GENERATOR>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>It is some configuration issue (or maybe a bug) in the SIP client 1 I think.</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff
 size=2>Once the call is established and the route has been learnt (using Record-Route), any new</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>requests should only be sent to the proxies on that learnt&nbsp;route.</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>Unfortunately I have heard that some SIP clients break this rule and send&nbsp;EVERY message to</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>the outbound proxy </FONT></SPAN><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>even when the&nbsp;outbound proxy did not add a Record-Route header.</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007>&gt;&gt;Could
 you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?</DEFANGHTML_SPAN></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>Of course, it should only get sent to proxy 2 if proxy 2 had added a Record-Route.</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>Like I say, to fix the problem, I think it's either a bug or configuration issue for SIP client 1.</FONT></SPAN></DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>  <DIV dir=ltr align=left><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>Regards,</FONT></SPAN></DIV><SPAN class=220540807-25072007><FONT face=Arial color=#0000ff size=2>  <DIV dir=ltr align=left><BR>Attila</DIV>  <DIV dir=ltr align=left></FONT></SPAN>&nbsp;</DIV>  <DIV dir=ltr align=left><SPAN
 class=220540807-25072007><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><BR>  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>  <HR tabIndex=-1>  <FONT face=Tahoma size=2><B>From:</B> sengottuvelan srirangan [mailto:sengottuvelan_s@yahoo.com] <BR><B>Sent:</B> 25 July 2007 06:15<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] Doubt about ACK and BYE <BR></FONT><BR></DIV>  <DIV></DIV>  <DIV>Hi SIP-experts,</DIV>  <DIV><FONT face=Arial color=#0000ff size=2></FONT>&nbsp;</DIV>  <DIV>I have doubt in a specific <DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Scenario where SIP
 client is set to send INVITE messages via outbound proxy. The setup is like below:</DEFANGHTML_SPAN></DIV>  <DIV><FONT face=Arial color=#0000ff size=2></FONT><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></DEFANGHTML_SPAN></DIV>  <DIV><FONT face=Arial color=#0000ff size=2></FONT><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">SIP-client 1---------------------------SIP outbound---------------------SIP proxy2---------------SIP client2</DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; 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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&amp;n! bsp;&nbsp;&nbsp;&nbsp;&nbsp;proxy1</DEFANGHTML_SPAN></DIV> 
 <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Here SIP client1 is registered&nbsp;with Proxy2 and set outbound proxy as proxy1. Outbound proxy1 is set not to include its ip address in the Record-Route header. After SIP 200 OK message is received in&nbsp;SIP-client1 which&nbsp;is still sending ACK and BYE messages via outbound proxy1 instead proxy2.</o:p></DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Could you anyone clarify that how can i send ACK and BYE messages to Registered proxy2 instead outbound proxy1?</o:p></DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY:
 Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <DIV><DEFANGHTML_SPAN s! tyle="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Regards</o:p></DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Sengottuvelan</o:p></DEFANGHTML_SPAN></DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <DIV><DEFANGHTML_SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></DEFANGHTML_SPAN>&nbsp;</DIV>  <div>  <HR SIZE=1>  Shape Yahoo! in your own image. <A href="http://us.rd.yahoo.com/evt=48517/*http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7">Join our Network Research Panel today!</A> </BLOCKQUOTE><BR><p>&#32;
      <hr size=1>Boardwalk for $500? In 2007? Ha! <br><a href="http://us.rd.yahoo.com/evt=48223/*http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow">Play Monopoly Here and Now</a> (it's updated for today's economy) at Yahoo! Games.
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Date: Wed, 25 Jul 2007 10:00:21 -0400
From: "Adam Fisk" <a@lastbamboo.org>
To: "Henry Sinnreich" <hsinnrei@adobe.com>
Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deploymentdatabefore LC
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We do plan on doing that eventually, Henry, but it will be awhile.  We're a
2 coder shop, so these things take us forever, but we will definitely do
that at some point.  It just might not be in 2007, honestly, so it would
likely be too late to be useful for the WG.

-Adam


On 7/24/07, Henry Sinnreich <hsinnrei@adobe.com> wrote:
>
>  Adam,
>
>
>
> I fully agree with your positive assessment of the ICE-17 document.
>
>
>
> Do you by any chance plan to also develop a test tool for testing the NAT
> environment and the success/failure cases, connection times, etc., to
> measure results?
>
>
>
> Or maybe you know someone who does?
>
>
>
> Thanks, Henry
>
>
>  ------------------------------
>
> *From:* adamfisk@gmail.com [mailto:adamfisk@gmail.com] *On Behalf Of *Adam
> Fisk
> *Sent:* Tuesday, July 24, 2007 3:08 PM
> *To:* Jiri Kuthan
> *Cc:* Henry Sinnreich; David Barrett; Jan Janak; Juha Heinanen; sip@ietf.org;
> mmusic@ietf.org
> *Subject:* Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
> deploymentdatabefore LC for RFC
>
>
>
> I've followed ICE development on the sidelines for years now, and I'm
> currently implementing ICE 17.  I've implemented what will be open source
> implementations of STUN and TURN, both client and server.  Here are the key
> points I've taken away from that experience:
>
> 1) The patience of Jonathan and everyone else in meticulously getting to
> the current draft is astonishing.
> 2) ICE is complex as hell and a pain in the ^#* to implement.  I was
> surprised at how hard it was, particularly for a draft that's so close to
> becoming an RFC.
> 3) ICE is solving a very tricky problem.
>
> I like Phillip's earlier suggestion that the draft stay in the "almost
> done" state like used to happen in the Routing group.  I realize that's
> asking a lot given how much work has gone into it and how eager everyone is
> to finalize it.  I think the problem is that standardizing NAT traversal is
> just hard, and hard problems are difficult to make simpler.  In my
> implementation of ICE, I do think it could be done more simply, but I don't
> have concrete suggestions right now because I'm still pulling my hair out
> with the implementation.  I'm not talking about drastic changes anywhere,
> but all the checklist fiddling sure takes a lot of patience to write.
> Simplifications might just be in the description and not necessarily the
> algorithm.
>
> I think it's a hard enough and important enough problem that we owe it to
> ourselves to be even more patient with an "almost done" state that adds the
> final touches based on real-world feedback.  I don't doubt for a second that
> ICE works, but it's so complex that I worry no one outside huge companies
> will ever implement it.  That would be a huge hindrance to ICE's adoption
> and power.
>
> -Adam
>

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We do plan on doing that eventually, Henry, but it will be awhile.&nbsp; We&#39;re a 2 coder shop, so these things take us forever, but we will definitely do that at some point.&nbsp; It just might not be in 2007, honestly, so it would likely be too late to be useful for the WG.
<br><br>-Adam<br><br><br><div><span class="gmail_quote">On 7/24/07, <b class="gmail_sendername">Henry Sinnreich</b> &lt;<a href="mailto:hsinnrei@adobe.com">hsinnrei@adobe.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">











<div link="blue" vlink="purple" lang="EN-US">

<div>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Adam,</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">I fully agree with your positive assessment
of the ICE-17 document.</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Do you by any chance plan to also develop
a test tool for testing the NAT environment and the success/failure cases, connection
times, etc., to measure results?</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Or maybe you know someone who does?</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Thanks, Henry</span></font></p>

<p><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">&nbsp;</span></font></p>

<div>

<div style="text-align: center;" align="center"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">

<hr align="center" size="2" width="100%">

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

<p><b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">
<a href="mailto:adamfisk@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">adamfisk@gmail.com</a> [mailto:<a href="mailto:adamfisk@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
adamfisk@gmail.com</a>] <b><span style="font-weight: bold;">On Behalf Of </span></b>Adam Fisk<br>
<b><span style="font-weight: bold;">Sent:</span></b> Tuesday, July 24, 2007 3:08
PM<br>
<b><span style="font-weight: bold;">To:</span></b> Jiri Kuthan<br>
<b><span style="font-weight: bold;">Cc:</span></b> Henry
 Sinnreich; David Barrett;
Jan Janak; Juha
 Heinanen; <a href="mailto:sip@ietf.o" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip@ietf.o</a>rg; <a href="mailto:mmusic@ietf.o" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
mmusic@ietf.o</a>rg<span class="q"><br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [MMUSIC] Re: [Sip]
RE: [BEHAVE] Re: ICE deploymentdatabefore LC for RFC</span></span></font></p>

</div>

<p><font face="Times New Roman" size="3"><span style="font-size: 12pt;">&nbsp;</span></font></p><div><span class="e" id="q_113f9ffcbfb919bd_3">

<p style="margin-bottom: 12pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;">I&#39;ve followed ICE
development on the sidelines for years now, and I&#39;m currently implementing ICE
17.&nbsp; I&#39;ve implemented what will be open source implementations of STUN and
TURN, both client and server.&nbsp; Here are the key points I&#39;ve taken away
from that experience: <br>
<br>
1) The patience of Jonathan and everyone else in meticulously getting to the
current draft is astonishing.<br>
2) ICE is complex as hell and a pain in the ^#* to implement.&nbsp; I was
surprised at how hard it was, particularly for a draft that&#39;s so close to
becoming an RFC. <br>
3) ICE is solving a very tricky problem.<br>
<br>
I like Phillip&#39;s earlier suggestion that the draft stay in the &quot;almost
done&quot; state like used to happen in the Routing group.&nbsp; I realize
that&#39;s asking a lot given how much work has gone into it and how eager everyone
is to finalize it.&nbsp; I think the problem is that standardizing NAT traversal
is just hard, and hard problems are difficult to make simpler.&nbsp; In my
implementation of ICE, I do think it could be done more simply, but I don&#39;t
have concrete suggestions right now because I&#39;m still pulling my hair out with
the implementation.&nbsp; I&#39;m not talking about drastic changes anywhere, but
all the checklist fiddling sure takes a lot of patience to write.&nbsp;
Simplifications might just be in the description and not necessarily the
algorithm. <br>
<br>
I think it&#39;s a hard enough and important enough problem that we owe it to
ourselves to be even more patient with an &quot;almost done&quot; state that
adds the final touches based on real-world feedback.&nbsp; I don&#39;t doubt for a
second that ICE works, but it&#39;s so complex that I worry no one outside huge
companies will ever implement it.&nbsp; That would be a huge hindrance to ICE&#39;s
adoption and power. <br>
<br>
-Adam</span></font></p>

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

</div>


</blockquote></div><br>

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From sip-bounces@ietf.org Wed Jul 25 14:50:29 2007
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To: IETF SIP List <sip@ietf.org>
From: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 25 Jul 2007 13:49:53 -0500
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Subject: [Sip] Ruminations on the SIP extension model
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There is a deep philosophical divide about how the SIP protocol  
should be formed and extended. One branch of this debate holds that  
SIP's role is as a transport protocol with a very few primitives and  
application logic handled at the data level. In this model, new  
applications can generally be built without changing the protocol or  
its syntax. The second branch holds that the SIP protocol itself  
should be intimately tied to the application, with new functionality  
represented as new message types. Yet another branch of the debate  
holds that SIP is only for setting up and tearing down "sessions",  
and that information outside of that needed to set up or tear down a  
"session" is rightfully part of the "session" and not in the domain  
of the SIP protocol itself.

Let's consider the user-stimulus over-SIP question (which is all  
mixed up with the emulating-DTMF-for-telephony question).

The "reusable primitive" camp might define a way to use an extensible  
mechanism (like SIP Events or a new INFO negotiation) and define user- 
stimulus delivery using a MIME object and use the extensible  
mechanism's negotiation feature to figure out whether a peer supports  
the usage (note that this is what we did with KPML).

The "extend SIP" camp might define a new SIP non-invite method type  
(maybe a method called "USERSTIM") that would carry a fixed payload  
and be negotiated using basic SIP feature negotiation mechanism (the  
"Allow" and "Supported" header field). One might note that once we  
have a few hundred SIP method types that these header fields are  
going to get relatively large.

The "it's just a session" camp might define a "user stimulus  
session", describe it using SDP, and use INVITE to establish a  
session over which user-stimulus information would be transmitted.

There has been a certain lack of clarity about the this extension  
model in SIP's past, resulting in some extensions that re-use a  
single SIP method (Event packages, INFO) and others that add new SIP  
methods (MESSAGE) and others that establish sessions (MSRP chat, Real- 
time text). It could be argued that this has made for a more complex  
protocol suite than might have occurred had any single methodology  
been followed exclusively.


It seems to me that we have a quandary -- if we accept the "Add a new  
method type for each application" model, then we should deprecate RFC  
3265, instead adding new SIP methods like SUBSCRIBE-KPML and NOTIFY- 
KPML. If we accept the "SIP as transport" model, then we have to ask  
if we need a way to negotiate application context for transported  
data within an INVITE-bounded dialog (like INFO "packages"). If we  
accept the "it's just a session" model then everything we've done  
with event packages, INFO, MESSAGE, and so needs to be replaced with  
a proper session-based mechanism (presence sessions, ISUP sessions,  
messaging sessions, and so on).

Or we could do nothing, accept that we have a rather muddled  
protocol, and just try and give guidance on when to use each  
extension modality.

Specifically, as the ongoing body-type-context debate has pointed  
out, while we can say what body types we support, we don't currently  
have a good way for a describe what a message containing a body of a  
specific type is likely to mean in the current context, and the jury  
is out as to whether we have reliable means of deducing a context.

We do have RFC 4485, which seems to me to do a fair job of guiding us  
on new header fields but doesn't say enough about body types (and  
doesn't even begin to discuss the body-type-context question beyond a  
vague mention of Content-Disposition), and which provides little  
useful guidance on when to use new methods vs. when to use events.

What do we want to do here? Is there a way to get a clear position,  
or are we going to be debating bodies vs. header fields vs. method- 
types forever?

--
Dean



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From sip-bounces@ietf.org Wed Jul 25 14:51:16 2007
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Subject: [Sip] INFO and what to do about it
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We have a clear consensus in the WG that "something needs to be done  
about INFO."

In this week's SIP meeting, we discussed a draft by Eric Burger that  
proposes formally restricting the usage of INFO to the existing RFCs  
that exercise it.

Several people noted that just making this limitation in a void is  
insufficient -- that we also need to explain the fundamental  
architectural problem and give guidance on how to more effectively  
solve the kinds of problems that people have been trying to solve  
using INFO.

We also have a contingent that thinks that we should fix info by  
adding a negotiation mechanism and registration procedure for INFO  
usages, much as SIP events have been done.

Others think that rather than an extensible mechanism using one  
message type (INFO), we should instead extend SIP as needed with new  
message types, further blurring the line between protocol and  
application (which I'll admit is already pretty blurry in SIP's  
case). Note that this is a follow-on mechanism to the first step of  
documenting NOT using INFO.

It seems to me that this means we need to clarify the SIP extension  
model, which I'll come back to later.

At this week's meeting, there was a majority (at least a 2 to 1  
ratio, and more by some counts) in favor of taking the path of Eric's  
draft and declaring a formal moratorium on new usages of INFO. This  
is pretty close to "rough consensus", but not so close that I was  
willing to declare consensus without taking the question to the list.

So here's a proposal:

I suggest we accept the model of Eric's draft, but ask him to add  
more depth to the architectural discussion and ask Christer to  
contribute some text in this direction, as he has some very definite  
ideas of what needs to be addressed. This leaves us with clear  
direction on not extending INFO with new usages or a negotiation  
mechanism.

Please discuss this on-list, especially if you object or have a  
counter-proposal.

This leaves open the question of clarifying the extension model of  
SIP, which I will raise in another thread to make following the  
discussion easier.

--
Dean Willis
SIP co-chair


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On 7/25/07 1:49 PM, Dean Willis wrote:
> It seems to me that we have a quandary -- if we accept the "Add a new 
> method type for each application" model, then we should deprecate RFC 
> 3265, instead adding new SIP methods like SUBSCRIBE-KPML and NOTIFY-KPML.

No, not really.

In pursuing such a model, you're effectively arguing that INVITE should 
be split into INVITE-AUDIO, INVITE-VIDEO, INVITE-AUDIO-VIDEO, etc. Why 
didn't we? Because pretty much all the SIP protocol semantics are 
identical regardless of what kind of INVITE session you're setting up.

Identically, just about all of the SIP protocol semantics are identical 
for SUBSCRIBE and NOTIFY, _regardless_ of the event package being used.

/a


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Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICEdeploymentdatabefore LC
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For those that may not know, another example of this is IAX2 which is
currently under review for publication by the IETF.  This protocol
clearly competes with SIP at least on some fronts.  It may be difficult
for new protocols to emerge, particularly with there is such huge
momentum behind one, but it can happen.

FM


-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org] 
Sent: Monday, July 23, 2007 2:38 PM
To: Henry Sinnreich; David Barrett; Jan Janak; Juha Heinanen
Cc: sip@ietf.org; mmusic@ietf.org
Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICEdeploymentdatabefore
LC for RFC

At 19:10 21/07/2007, Henry Sinnreich wrote:
>I fully agree with Jiri that this is an IETF problem and not an ICE
>problem.
>ICE-17 is a great document, no doubt about it.
>
>What happens however of someone finds an innovative approach that may
>work better? Will it be rejected because it will not be RFC standards
>compliant?

I don't think so, IETF is having a history of working on competing
standards,
jabber and simple being a prominent and recent example thereof. I think
that's
actually with current IETF's size and dynamics a fairly good strategy to
eliminate single points of failure. 

-jiri


>What could an innovator tell customers about a solution that works
>better but is not IETF ICE RFC standards compliant?
>
>Henry
>
>-----Original Message-----
>From: Jiri Kuthan [mailto:jiri@iptel.org] 
>Sent: Saturday, July 21, 2007 9:48 AM
>To: David Barrett; 'Jan Janak'; 'Juha Heinanen'
>Cc: sip@ietf.org; mmusic@ietf.org
>Subject: RE: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE
>deploymentdatabefore LC for RFC
>
>At 20:05 19/07/2007, David Barrett wrote:
>>> -----Original Message-----
>>> From: Jiri Kuthan [mailto:jiri@iptel.org]
>>> Subject: Re: [MMUSIC] Re: [Sip] RE: [BEHAVE] Re: ICE deployment
>databefore
>>> LC for RFC
>>> 
>>> So I think that the right course of action is that folks doubting
ICE
>>> reliability
>>> make their case (by that I mean specific arguments rather than
>emotional
>>> debates
>>> about red-colored nails and other hardware).
>>
>>Er, my "specific argument" is "until it's been proven to work, it
>hasn't
>>been proven to work".  It's tautological, but no less true for it.
>>
>>Indeed, if you're *not* doubting ICE, why aren't you?
>
>Hi David,
>
>I'm not entirely sure what is the step you are suggesting?
>
>I think ICE is architecturally right answer to the right problem.
>Notwithstanding that, there are certainly some things I prefer
>differently (e.g., the infamous debate about proactive hole punching)
>or may turn out to be bugs. I think though that the key problem at
>this point is of institutional nature. In the IETF language, "hasn't
>been proven to work" is adequate to "lacking running code" and this
>appears to be a phenomen which is indeed occuring in the IETF and
>is not unique to ICE.
>
>The question is how to get this fixed?
>
>Some other standardization bodies enjoying the amount of contributions
>and complexity as the IETF is enjoying now (S in SIP does not really
>stand for Simple), proceed with 'after-fact filters', in that they
>produce the initial specs faster and later produce 'profiles' which
>separate what has proved and what less. Plus -bis versions are produced
>for whatever appears worth puttting effort in it (i.e., for which there
>is running code and field evidence of improvement need).This could be 
>an approach  which may be more feasible in today's IETF's dimension
than
>
>trying to get back to historical roots.
>
>-jiri 
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip



--
Jiri Kuthan            http://iptel.org/~jiri/



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



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Date: Thu, 26 Jul 2007 15:09:17 +1000
From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: "vishal jogi" <vishaljogi@gmail.com>
Subject: Re: [Sip] Event - list subscribe
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This is incorrect behaviour by the RLS. The list certainly exists. It just
happens to be empty. A 2xx response to the SUBSCRIBE followed by a NOTIFY
with the appropriate payload is what I would consider to be the correct
behaviour.

Thanks,
Hisham


On 19/07/07, vishal jogi <vishaljogi@gmail.com> wrote:
>
> Hi,
> I have one doubt.
>
> i m subscribing to RLS uri,
> which is refering one Resource-list which doesn't have any <entry>
> mean it's blank <LIST>
>
> server is sending 404 as response for Subscribe, can anyone tell me it's
> correct behaviour of server,
> as that <LIST> is existing
>
> Thanks,
> Vishal...
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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<div>This is incorrect behaviour by the RLS. The list certainly exists. It just happens to be empty. A 2xx response to the SUBSCRIBE followed by a NOTIFY with the appropriate payload is what I would consider to be the correct behaviour.
</div>
<div>&nbsp;</div>
<div>Thanks,</div>
<div>Hisham<br><br>&nbsp;</div>
<div><span class="gmail_quote">On 19/07/07, <b class="gmail_sendername">vishal jogi</b> &lt;<a href="mailto:vishaljogi@gmail.com">vishaljogi@gmail.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>Hi,</div>
<div>I have one doubt.</div>
<div>&nbsp;</div>
<div>i m subscribing to RLS uri,</div>
<div>which is refering one Resource-list which doesn&#39;t have any &lt;entry&gt;</div>
<div>mean it&#39;s blank &lt;LIST&gt;</div>
<div>&nbsp;</div>
<div>server is sending 404 as response for Subscribe, can anyone tell me it&#39;s correct behaviour of server, </div>
<div>as that &lt;LIST&gt; is existing</div>
<div>&nbsp;</div>
<div>Thanks,</div><span class="sg">
<div>Vishal...</div></span><br>_______________________________________________<br>Sip mailing list &nbsp;<a onclick="return top.js.OpenExtLink(window,event,this)" href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank">
https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu
</a> for questions on current sip<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br></blockquote></div><br>

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From sip-bounces@ietf.org Thu Jul 26 02:48:14 2007
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Subject: RE: [Sip] Ruminations on the SIP extension model
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Dean,

I think you're looking at this all wrong. I believe your "camps" below
don't reflect what's really going on.

Most people who develop using SIP do not have a full understanding of
how it really works, nor do they have time to figure it out. They have a
stack that gives them some APIs of varying granularity that they must
use to create the feature they need to make their deadlines. Some get it
right, some get overwhelmed by the complexity of the protocol and make a
mistake, others don't care as long as their test plans execute
successfully. Generally speaking the philosophy they employ "the path of
least (or near-least) resistance".

INFO for carrying user-stimulus/DTMF is a great example of this. I can
pretty much tell you what happened to get us to where we are today.

Designer A: I need to carry some DTMF digits from the client to my
voicemail server so I can enter my password. I don't want to mess with
RTP because that's harder to do, so I'll do it in SIP since it's easier
to code for me. All I need is a quickie in-band message to the gateway
to tell it to play some tone and I'll let the gateway guy who has to
deal with RTP and codecs anyhow figure it out from there. I'll use INFO
since it's handy. Nobody has defined this that I know of (or I have to
work with some legacy device) so I'll create my own message body type
and just throw it in there. If someone doesn't understand it they'll
just drop it. All of the devices I have to test with for this feature
(today) have agreed to support this mechanism so I'm golden. Besides,
now it's in SIP so I can have my proxy do things with the digits in the
future if I want.

Designer B: I need to carry some DTMF digits from the client to my
voicemail server so I can enter my password. I already have code that
handles voice packets, so rather than try to mess with all of that crazy
SIP stuff, I'll just jam something into the RTP stream to tell the other
end to play DTMF tone X. I'll use RFC-2833. If someone doesn't
understand it they'll just drop it. Nobody needs to route off of these
digits, so there's no risk in not carrying this in SIP. All of the
devices I have to test with for this feature (today) have agreed to
support this mechanism so I'm golden.

Repeat designer A 2-4 times in semi-parallel, and throw a designer B in
there and you have what we have today for DTMF.

Contrast this with RFC-3265. Not only did it describe a general
mechanism, but it clearly lays out a way forward for new applications
that is simple and understandable to the average reader. The mechanism
is widely supported in SIP stack APIs. Designer A will use it because
it's already there and they can figure it out (although what goes into
the message bodies may still vary somewhat). Designer B will use it too
because their device probably has an easier to use API these days to
make it happen. It's the path of least resistance and it's a good
mechanism, so we win.

You don't see INVITE-AUDIO or SUBSCRIBE-KPML because it's
counter-intuitive and more difficult to implement. INFO is easy to
understand, it's sitting there in their stack API, and they don't have
to do too much to make it work. You put bits in one end and they pop out
the other end, no muss, no fuss.

Metaphorically speaking, applications are nouns, not verbs. Protocols
only need to contain a very rudimentary set of verbs to facilitate
transporting all the rest of the 'parts of speech' so to speak (HTTP is
another great example of this at work).

The real problem is that if we tell people what not to do (ala INFO)
without giving them an easy way of doing it correctly, then they're very
likely to ignore the advice given to them. Again, this is where I think
3265 got it right.

Regards,
Brian

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Wednesday, July 25, 2007 1:50 PM
> To: IETF SIP List
> Subject: [Sip] Ruminations on the SIP extension model
>=20
> There is a deep philosophical divide about how the SIP=20
> protocol should be formed and extended. One branch of this=20
> debate holds that SIP's role is as a transport protocol with=20
> a very few primitives and application logic handled at the=20
> data level. In this model, new applications can generally be=20
> built without changing the protocol or its syntax. The second=20
> branch holds that the SIP protocol itself should be=20
> intimately tied to the application, with new functionality=20
> represented as new message types. Yet another branch of the=20
> debate holds that SIP is only for setting up and tearing down=20
> "sessions", and that information outside of that needed to=20
> set up or tear down a "session" is rightfully part of the=20
> "session" and not in the domain of the SIP protocol itself.
>=20
> Let's consider the user-stimulus over-SIP question (which is=20
> all mixed up with the emulating-DTMF-for-telephony question).
>=20
> The "reusable primitive" camp might define a way to use an=20
> extensible mechanism (like SIP Events or a new INFO=20
> negotiation) and define user- stimulus delivery using a MIME=20
> object and use the extensible mechanism's negotiation feature=20
> to figure out whether a peer supports the usage (note that=20
> this is what we did with KPML).
>=20
> The "extend SIP" camp might define a new SIP non-invite=20
> method type (maybe a method called "USERSTIM") that would=20
> carry a fixed payload and be negotiated using basic SIP=20
> feature negotiation mechanism (the "Allow" and "Supported"=20
> header field). One might note that once we have a few hundred=20
> SIP method types that these header fields are going to get=20
> relatively large.
>=20
> The "it's just a session" camp might define a "user stimulus=20
> session", describe it using SDP, and use INVITE to establish=20
> a session over which user-stimulus information would be transmitted.
>=20
> There has been a certain lack of clarity about the this=20
> extension model in SIP's past, resulting in some extensions=20
> that re-use a single SIP method (Event packages, INFO) and=20
> others that add new SIP methods (MESSAGE) and others that=20
> establish sessions (MSRP chat, Real- time text). It could be=20
> argued that this has made for a more complex protocol suite=20
> than might have occurred had any single methodology been=20
> followed exclusively.
>=20
>=20
> It seems to me that we have a quandary -- if we accept the=20
> "Add a new method type for each application" model, then we=20
> should deprecate RFC 3265, instead adding new SIP methods=20
> like SUBSCRIBE-KPML and NOTIFY- KPML. If we accept the "SIP=20
> as transport" model, then we have to ask if we need a way to=20
> negotiate application context for transported data within an=20
> INVITE-bounded dialog (like INFO "packages"). If we accept=20
> the "it's just a session" model then everything we've done=20
> with event packages, INFO, MESSAGE, and so needs to be=20
> replaced with a proper session-based mechanism (presence=20
> sessions, ISUP sessions, messaging sessions, and so on).
>=20
> Or we could do nothing, accept that we have a rather muddled=20
> protocol, and just try and give guidance on when to use each=20
> extension modality.
>=20
> Specifically, as the ongoing body-type-context debate has=20
> pointed out, while we can say what body types we support, we=20
> don't currently have a good way for a describe what a message=20
> containing a body of a specific type is likely to mean in the=20
> current context, and the jury is out as to whether we have=20
> reliable means of deducing a context.
>=20
> We do have RFC 4485, which seems to me to do a fair job of=20
> guiding us on new header fields but doesn't say enough about=20
> body types (and doesn't even begin to discuss the=20
> body-type-context question beyond a vague mention of=20
> Content-Disposition), and which provides little useful=20
> guidance on when to use new methods vs. when to use events.
>=20
> What do we want to do here? Is there a way to get a clear=20
> position, or are we going to be debating bodies vs. header=20
> fields vs. method- types forever?
>=20
> --
> Dean
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: [Sip] comments on draft-munakata-sip-privacy-new
Date: Thu, 26 Jul 2007 13:12:01 +0200
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From: "Jesske, R" <R.Jesske@t-com.net>
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Dear Mayumi et all,

Your draft says:

8.  Proxy Behavior

   When a proxy receives a SIP message containing a privacy flag, the
   proxy compliant with this specification MUST NOT add any information
   that may reveal something about the sender that is irrelevant to
   routing unless the proxy knows that such information will be deleted
   before it leaves the boundary of the Trust Domain[RFC3324].

   A proxy MUST NOT modify the privacy flag, if present.


Does this text allows networks that are due to regulation which have to =
provide at least within the trusted Domain a Asserted Identity?=20
It looks that it is so. Could we spell out this explicitly?
And we should refer to RFC 3325 as already mentioned in an other mail.

What we should also to take into consideration is the case of ACR.
As you wrote already in Section 7.2 a indication showing that the =
economisation of the user identity were done at least by the UAC.
If the network provides ACR at least there must be a mechanism to =
identity such cases too.=20

I think we should at least mention the relation to this mechanism. And =
what could be done to fulfil the requirements of ACR.

Best Regards

Roland


Deutsche Telekom AG=20
T-Com, Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE33-6=20
Deutsche-Telekom-Allee 1, 64295 Darmstadt, Raum 314(Besucheradresse)=20
Postfach, 64307 Darmstadt (Postanschrift)=20
06151 83-5940 (Tel.)=20
06151 83-4577 (Fax)=20
E-Mail: r.jesske@t-com.net=20
http://www.t-com.de

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Dr. Klaus Zumwinkel (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender), Dr. Karl-Gerhard Eick =
(stellvertretender Vorsit-zender), Hamid Akhavan, Timotheus H=F6ttges, =
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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Ruminations on the SIP extension model
Date: Thu, 26 Jul 2007 08:59:59 -0500
To: Adam Roach <adam@nostrum.com>
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On Jul 25, 2007, at 1:57 PM, Adam Roach wrote:

> On 7/25/07 1:49 PM, Dean Willis wrote:
>> It seems to me that we have a quandary -- if we accept the "Add a  
>> new method type for each application" model, then we should  
>> deprecate RFC 3265, instead adding new SIP methods like SUBSCRIBE- 
>> KPML and NOTIFY-KPML.
>
> No, not really.
>
> In pursuing such a model, you're effectively arguing that INVITE  
> should be split into INVITE-AUDIO, INVITE-VIDEO, INVITE-AUDIO- 
> VIDEO, etc. Why didn't we? Because pretty much all the SIP protocol  
> semantics are identical regardless of what kind of INVITE session  
> you're setting up.
>
> Identically, just about all of the SIP protocol semantics are  
> identical for SUBSCRIBE and NOTIFY, _regardless_ of the event  
> package being used.

Well, personally I'm in the "reuseable primitives" camp, and I  
believe you are too (as 3265 is a shining example of this approach).  
But other people have very different beliefs.

Note that at least one person suggested at the mic during the  
discussion of Eric's draft that each different thing that is likely  
to be done with INFO should instead be done with a different new  
method "per thing".

Note also that one of the main contributors prior to RFC 2543 broke  
with the community over the BYE method, as he (IIRC) believed that  
the way to end a session was to stop sending media.

--
Dean



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Subject: RE: [Sip] Ruminations on the SIP extension model
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To top it off, the designers you describe might have actually
read RFC 2976 which states the following "example uses" for
INFO:
      - Carrying DTMF digits generated during a SIP session.=20

> -----Original Message-----
> From: Stucker, Brian (RICH1:AR00)=20
> Sent: Wednesday, July 25, 2007 23:47
> To: Dean Willis; IETF SIP List
> Subject: RE: [Sip] Ruminations on the SIP extension model
>=20
> Dean,
>=20
> I think you're looking at this all wrong. I believe your=20
> "camps" below don't reflect what's really going on.
>=20
> Most people who develop using SIP do not have a full=20
> understanding of how it really works, nor do they have time=20
> to figure it out. They have a stack that gives them some APIs=20
> of varying granularity that they must use to create the=20
> feature they need to make their deadlines. Some get it right,=20
> some get overwhelmed by the complexity of the protocol and=20
> make a mistake, others don't care as long as their test plans=20
> execute successfully. Generally speaking the philosophy they=20
> employ "the path of least (or near-least) resistance".
>=20
> INFO for carrying user-stimulus/DTMF is a great example of=20
> this. I can pretty much tell you what happened to get us to=20
> where we are today.
>=20
> Designer A: I need to carry some DTMF digits from the client=20
> to my voicemail server so I can enter my password. I don't=20
> want to mess with RTP because that's harder to do, so I'll do=20
> it in SIP since it's easier to code for me. All I need is a=20
> quickie in-band message to the gateway to tell it to play=20
> some tone and I'll let the gateway guy who has to deal with=20
> RTP and codecs anyhow figure it out from there. I'll use INFO=20
> since it's handy. Nobody has defined this that I know of (or=20
> I have to work with some legacy device) so I'll create my own=20
> message body type and just throw it in there. If someone=20
> doesn't understand it they'll just drop it. All of the=20
> devices I have to test with for this feature
> (today) have agreed to support this mechanism so I'm golden.=20
> Besides, now it's in SIP so I can have my proxy do things=20
> with the digits in the future if I want.
>=20
> Designer B: I need to carry some DTMF digits from the client=20
> to my voicemail server so I can enter my password. I already=20
> have code that handles voice packets, so rather than try to=20
> mess with all of that crazy SIP stuff, I'll just jam=20
> something into the RTP stream to tell the other end to play=20
> DTMF tone X. I'll use RFC-2833. If someone doesn't understand=20
> it they'll just drop it. Nobody needs to route off of these=20
> digits, so there's no risk in not carrying this in SIP. All=20
> of the devices I have to test with for this feature (today)=20
> have agreed to support this mechanism so I'm golden.
>=20
> Repeat designer A 2-4 times in semi-parallel, and throw a=20
> designer B in there and you have what we have today for DTMF.
>=20
> Contrast this with RFC-3265. Not only did it describe a=20
> general mechanism, but it clearly lays out a way forward for=20
> new applications that is simple and understandable to the=20
> average reader. The mechanism is widely supported in SIP=20
> stack APIs. Designer A will use it because it's already there=20
> and they can figure it out (although what goes into the=20
> message bodies may still vary somewhat). Designer B will use=20
> it too because their device probably has an easier to use API=20
> these days to make it happen. It's the path of least=20
> resistance and it's a good mechanism, so we win.
>=20
> You don't see INVITE-AUDIO or SUBSCRIBE-KPML because it's=20
> counter-intuitive and more difficult to implement. INFO is=20
> easy to understand, it's sitting there in their stack API,=20
> and they don't have to do too much to make it work. You put=20
> bits in one end and they pop out the other end, no muss, no fuss.
>=20
> Metaphorically speaking, applications are nouns, not verbs.=20
> Protocols only need to contain a very rudimentary set of=20
> verbs to facilitate transporting all the rest of the 'parts=20
> of speech' so to speak (HTTP is another great example of this=20
> at work).
>=20
> The real problem is that if we tell people what not to do=20
> (ala INFO) without giving them an easy way of doing it=20
> correctly, then they're very likely to ignore the advice=20
> given to them. Again, this is where I think
> 3265 got it right.
>=20
> Regards,
> Brian
>=20
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: Wednesday, July 25, 2007 1:50 PM
> > To: IETF SIP List
> > Subject: [Sip] Ruminations on the SIP extension model
> >=20
> > There is a deep philosophical divide about how the SIP=20
> protocol should=20
> > be formed and extended. One branch of this debate holds that SIP's=20
> > role is as a transport protocol with a very few primitives and=20
> > application logic handled at the data level. In this model, new=20
> > applications can generally be built without changing the=20
> protocol or=20
> > its syntax. The second branch holds that the SIP protocol itself=20
> > should be intimately tied to the application, with new=20
> functionality=20
> > represented as new message types. Yet another branch of the debate=20
> > holds that SIP is only for setting up and tearing down=20
> "sessions", and=20
> > that information outside of that needed to set up or tear down a=20
> > "session" is rightfully part of the "session" and not in=20
> the domain of=20
> > the SIP protocol itself.
> >=20
> > Let's consider the user-stimulus over-SIP question (which=20
> is all mixed=20
> > up with the emulating-DTMF-for-telephony question).
> >=20
> > The "reusable primitive" camp might define a way to use an=20
> extensible=20
> > mechanism (like SIP Events or a new INFO
> > negotiation) and define user- stimulus delivery using a MIME object=20
> > and use the extensible mechanism's negotiation feature to=20
> figure out=20
> > whether a peer supports the usage (note that this is what=20
> we did with=20
> > KPML).
> >=20
> > The "extend SIP" camp might define a new SIP non-invite method type=20
> > (maybe a method called "USERSTIM") that would carry a fixed payload=20
> > and be negotiated using basic SIP feature negotiation=20
> mechanism (the=20
> > "Allow" and "Supported"
> > header field). One might note that once we have a few hundred SIP=20
> > method types that these header fields are going to get relatively=20
> > large.
> >=20
> > The "it's just a session" camp might define a "user=20
> stimulus session",=20
> > describe it using SDP, and use INVITE to establish a session over=20
> > which user-stimulus information would be transmitted.
> >=20
> > There has been a certain lack of clarity about the this extension=20
> > model in SIP's past, resulting in some extensions that=20
> re-use a single=20
> > SIP method (Event packages, INFO) and others that add new=20
> SIP methods=20
> > (MESSAGE) and others that establish sessions (MSRP chat, Real- time=20
> > text). It could be argued that this has made for a more complex=20
> > protocol suite than might have occurred had any single methodology=20
> > been followed exclusively.
> >=20
> >=20
> > It seems to me that we have a quandary -- if we accept the=20
> > "Add a new method type for each application" model, then we=20
> > should deprecate RFC 3265, instead adding new SIP methods=20
> > like SUBSCRIBE-KPML and NOTIFY- KPML. If we accept the "SIP=20
> > as transport" model, then we have to ask if we need a way to=20
> > negotiate application context for transported data within an=20
> > INVITE-bounded dialog (like INFO "packages"). If we accept=20
> > the "it's just a session" model then everything we've done=20
> > with event packages, INFO, MESSAGE, and so needs to be=20
> > replaced with a proper session-based mechanism (presence=20
> > sessions, ISUP sessions, messaging sessions, and so on).
> >=20
> > Or we could do nothing, accept that we have a rather muddled=20
> > protocol, and just try and give guidance on when to use each=20
> > extension modality.
> >=20
> > Specifically, as the ongoing body-type-context debate has=20
> > pointed out, while we can say what body types we support, we=20
> > don't currently have a good way for a describe what a message=20
> > containing a body of a specific type is likely to mean in the=20
> > current context, and the jury is out as to whether we have=20
> > reliable means of deducing a context.
> >=20
> > We do have RFC 4485, which seems to me to do a fair job of=20
> > guiding us on new header fields but doesn't say enough about=20
> > body types (and doesn't even begin to discuss the=20
> > body-type-context question beyond a vague mention of=20
> > Content-Disposition), and which provides little useful=20
> > guidance on when to use new methods vs. when to use events.
> >=20
> > What do we want to do here? Is there a way to get a clear=20
> > position, or are we going to be debating bodies vs. header=20
> > fields vs. method- types forever?
> >=20
> > --
> > Dean
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] Ruminations on the SIP extension model
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> 
> Well, personally I'm in the "reuseable primitives" camp, and I
> believe you are too (as 3265 is a shining example of this approach).
> But other people have very different beliefs.

I think everyone would be for reusable primitives.  The concern I have,
though, is that the problem with the currently defined primitive for dtmf is
it is the wrong type for the dtmf feature.  A second concern is that the
defined primitive is far too heavy for the problem being addressed.  So my
marketing gut tells me that market forces will continue to work against it.
 
> Note also that one of the main contributors prior to RFC 2543 broke
> with the community over the BYE method, as he (IIRC) believed that
> the way to end a session was to stop sending media.

Funny.  I was thinking it should be a re-Invite with an expires=0 and sdp
media ports of 0.
;)

-hadriel



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Subject: RE: [Sip] Ruminations on the SIP extension model
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> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 26 July 2007 16:17
> To: 'Dean Willis'; 'Adam Roach'
> Cc: 'IETF SIP List'
> Subject: RE: [Sip] Ruminations on the SIP extension model
>=20
>=20
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> >=20
> > Note also that one of the main contributors prior to RFC 2543 broke=20
> > with the community over the BYE method, as he (IIRC)=20
> believed that the=20
> > way to end a session was to stop sending media.
>=20
> Funny.  I was thinking it should be a re-Invite with an=20
> expires=3D0 and sdp media ports of 0.
> ;)
>=20
> -hadriel

Nonsense. The initial INVITE should cause an implicit subscription for a
bye event notification. But I guess we have to use the "legacy" BYE
method :)



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Subject: Re: [Sip] Ruminations on the SIP extension model
Date: Thu, 26 Jul 2007 10:26:04 -0500
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If any of this shows up at a SIPit, you owe me beer. And asprin.

RjS

On Jul 26, 2007, at 10:23 AM, Steve Langstaff wrote:

>> -----Original Message-----
>> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
>> Sent: 26 July 2007 16:17
>> To: 'Dean Willis'; 'Adam Roach'
>> Cc: 'IETF SIP List'
>> Subject: RE: [Sip] Ruminations on the SIP extension model
>>
>>
>>> -----Original Message-----
>>> From: Dean Willis [mailto:dean.willis@softarmor.com]
>>>
>>> Note also that one of the main contributors prior to RFC 2543 broke
>>> with the community over the BYE method, as he (IIRC)
>> believed that the
>>> way to end a session was to stop sending media.
>>
>> Funny.  I was thinking it should be a re-Invite with an
>> expires=0 and sdp media ports of 0.
>> ;)
>>
>> -hadriel
>
> Nonsense. The initial INVITE should cause an implicit subscription  
> for a
> bye event notification. But I guess we have to use the "legacy" BYE
> method :)
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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On 7/26/07 8:59 AM, Dean Willis wrote:
> Well, personally I'm in the "reuseable primitives" camp, and I believe 
> you are too (as 3265 is a shining example of this approach). But other 
> people have very different beliefs.


Actually, if I knew then what I knew now, I'd probably have proposed 
doing things completely differently. I have come to believe that a far 
cleaner approach would have been to use INVITE to set up a stream of 
state change notifications -- it would have prevented this "multiple 
usages in a dialog" mess; it would have prevented the bizarre forking 
SUBSCRIBE problem; it would have made unsolicited NOTIFYs impossible... 
the whole setup would have been much cleaner.

And, as we all know by now, every time you send an unsolicited NOTIFY, 
the terrorists win.

/a


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Subject: [Sip] ICE tutorial in Chicago  - PLEASE PAY IF YOU ATE LUNCH
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Lunch during the tutorial was paid out of my pocket, and I was expecting 
to get re-imbursed by everyone that took a sandwich. Though all of the 
food was taken by the end of the tutorial, I am several hundred dollars 
short. Please, if you took lunch and did not give me $20, give it to me 
before you leave, and if you have not, contact me privately and we can 
arrange for paper mail.

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



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Dean Willis wrote:
> There is a deep philosophical divide about how the SIP protocol should 
> be formed and extended. One branch of this debate holds that SIP's role 
> is as a transport protocol with a very few primitives and application 
> logic handled at the data level. In this model, new applications can 
> generally be built without changing the protocol or its syntax. The 
> second branch holds that the SIP protocol itself should be intimately 
> tied to the application, with new functionality represented as new 
> message types. Yet another branch of the debate holds that SIP is only 
> for setting up and tearing down "sessions", and that information outside 
> of that needed to set up or tear down a "session" is rightfully part of 
> the "session" and not in the domain of the SIP protocol itself.

I would argue that SIP *is* an application protocol - supporting 
'multimedia communications'. Its implementation takes many forms - 
softphones, hardphones, gateways, and so on, but still it is all about 
multimedia communications.

Consequently, the notion of an additional layer *above* SIP, something 
beyond the UI, something that has behaviors that are fundamentally 
different (and NOT compatible) with other applications that might reside 
above SIP, is foreign to the design of SIP, and something that is 
dangerous. It is this thinking that will lead us to highly segmented 
uses of SIP, none of which are well defined or interoperable. Very 
reminiscent of the 'service ID' discussions in sipping, in fact.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: "Adam Fisk" <a@lastbamboo.org>
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On 7/24/07, Jiri Kuthan <jiri@iptel.org> wrote:
>
> At 22:07 24/07/2007, Adam Fisk wrote:
> >I've followed ICE development on the sidelines for years now, and I'm
> currently implementing ICE 17.  I've implemented what will be open source
> implementations of STUN and TURN, both client and server.
>
> That's marvellous news -- can you send me a pointer to that?


We're not quite public yet, Jiri.  I'll send a note to the list when we're
up, though.

-Adam

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<br><br><div><span class="gmail_quote">On 7/24/07, <b class="gmail_sendername">Jiri Kuthan</b> &lt;<a href="mailto:jiri@iptel.org">jiri@iptel.org</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
At 22:07 24/07/2007, Adam Fisk wrote:<br>&gt;I&#39;ve followed ICE development on the sidelines for years now, and I&#39;m currently implementing ICE 17.&nbsp;&nbsp;I&#39;ve implemented what will be open source implementations of STUN and TURN, both client and server.
<br><br>That&#39;s marvellous news -- can you send me a pointer to that?</blockquote></div><br>We&#39;re not quite public yet, Jiri.&nbsp; I&#39;ll send a note to the list when we&#39;re up, though.&nbsp; <br><br>-Adam<br>

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_______________________________________________
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From sip-bounces@ietf.org Thu Jul 26 17:03:42 2007
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Thread-Topic: draft-ietf-sip-sips-05: 480 vs. 418
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Subject: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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In the meeting, the issue of the error code to be used when the wrong
URI is
used in a request-URI (i.e., sip instead of sips, or vice-versa) came
up.

The status quo is 2 new error codes (418 and 419). Another proposal was
to use only one
(i.e., 418 with 2 new headers).

That proposal did not seem to get much traction. And people were not
very with the
status quo either.=20

Another proposal was to use 480 (Temporarily Unavailable) and NOT to use
an explicit
indication of the required/supported URIs (SIP, SIPS, etc.). Of course,
one can always
put text in the reason phrase to describe the error for displaying to
the end-use.

After talking to all the individual who seemed to have strong opinions
on this topic,
it appears that the 480 approach is the preferred approach.=20

Please let the list know if you are strongly opposed to this outcome,
along with the=20
reasons behind your opposition.

This is the last remaining issue to be closed in the draft, as a result
of WGLC.


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At 04:03 PM 7/26/2007, Francois Audet wrote:
>In the meeting, the issue of the error code to be used when the wrong URI is
>used in a request-URI (i.e., sip instead of sips, or vice-versa) came up.
>
>The status quo is 2 new error codes (418 and 419).

(BTW 419 is used by Cullen's hashcash draft)

>Another proposal was to use only one (i.e., 418 with 2 new headers).

I would advocate this proposal, especially verses a 480, as a (new) 
Reason code to give the specifics can easily be stripped along the 
way back to the UAC, leaving it clueless as to what the actual 
problem was. Also not all implementations support Reason (though they 
should).

A 418 is specific to the point of what was wrong.... and all UACs 
have to deal with the fact they can become a 400 (thus removing the details)


>That proposal did not seem to get much traction. And people were not
>very with the status quo either.
>
>Another proposal was to use 480 (Temporarily Unavailable) and NOT to use
>an explicit indication of the required/supported URIs (SIP, SIPS, 
>etc.). Of course,
>one can always put text in the reason phrase to describe the error 
>for displaying to
>the end-use.
>
>After talking to all the individual who seemed to have strong opinions
>on this topic, it appears that the 480 approach is the preferred approach.
>
>Please let the list know if you are strongly opposed to this outcome,
>along with the reasons behind your opposition.
>
>This is the last remaining issue to be closed in the draft, as a result
>of WGLC.
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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



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From: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Thu, 26 Jul 2007 16:49:35 -0500
To: "Francois Audet" <audet@nortel.com>
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I think the 480 approach described below is the right thing to do.

RjS

On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:

> In the meeting, the issue of the error code to be used when the wrong
> URI is
> used in a request-URI (i.e., sip instead of sips, or vice-versa) came
> up.
>
> The status quo is 2 new error codes (418 and 419). Another proposal  
> was
> to use only one
> (i.e., 418 with 2 new headers).
>
> That proposal did not seem to get much traction. And people were not
> very with the
> status quo either.
>
> Another proposal was to use 480 (Temporarily Unavailable) and NOT  
> to use
> an explicit
> indication of the required/supported URIs (SIP, SIPS, etc.). Of  
> course,
> one can always
> put text in the reason phrase to describe the error for displaying to
> the end-use.
>
> After talking to all the individual who seemed to have strong opinions
> on this topic,
> it appears that the 480 approach is the preferred approach.
>
> Please let the list know if you are strongly opposed to this outcome,
> along with the
> reasons behind your opposition.
>
> This is the last remaining issue to be closed in the draft, as a  
> result
> of WGLC.
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org Thu Jul 26 18:21:59 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Thu, 26 Jul 2007 17:21:39 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "James M. Polk" <jmpolk@cisco.com>, "IETF SIP List" <sip@ietf.org>
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Hi James,

The specific reason why many were advocating a 480 instead of a new
error code is exactly because they did NOT want the error code=20
to encourage UAC to reattempt the request by downgrading to
SIP automatically.

This is the point that Eric Rescorla was making in the meeting, and
it has been re-iterated in one way or another by both Jon Peterson
and by Rohan Mahy.

You *could* of course with this approach use a reason phrase in the
480 to provide a message to the user of what the issue is.=20

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Thursday, July 26, 2007 14:31
> To: Audet, Francois (SC100:3055); IETF SIP List
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> At 04:03 PM 7/26/2007, Francois Audet wrote:
> >In the meeting, the issue of the error code to be used when=20
> the wrong=20
> >URI is used in a request-URI (i.e., sip instead of sips, or=20
> vice-versa) came up.
> >
> >The status quo is 2 new error codes (418 and 419).
>=20
> (BTW 419 is used by Cullen's hashcash draft)
>=20
> >Another proposal was to use only one (i.e., 418 with 2 new headers).
>=20
> I would advocate this proposal, especially verses a 480, as a=20
> (new) Reason code to give the specifics can easily be=20
> stripped along the way back to the UAC, leaving it clueless=20
> as to what the actual problem was. Also not all=20
> implementations support Reason (though they should).
>=20
> A 418 is specific to the point of what was wrong.... and all=20
> UACs have to deal with the fact they can become a 400 (thus=20
> removing the details)
>=20
>=20
> >That proposal did not seem to get much traction. And people were not
> >very with the status quo either.
> >
> >Another proposal was to use 480 (Temporarily Unavailable)=20
> and NOT to use
> >an explicit indication of the required/supported URIs (SIP, SIPS,=20
> >etc.). Of course,
> >one can always put text in the reason phrase to describe the error=20
> >for displaying to
> >the end-use.
> >
> >After talking to all the individual who seemed to have=20
> strong opinions
> >on this topic, it appears that the 480 approach is the=20
> preferred approach.
> >
> >Please let the list know if you are strongly opposed to this outcome,
> >along with the reasons behind your opposition.
> >
> >This is the last remaining issue to be closed in the draft,=20
> as a result
> >of WGLC.
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Thu Jul 26 18:37:44 2007
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Date: Thu, 26 Jul 2007 23:37:36 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Francois,

I am reasonably happy with this proposal, BUT I must point out the
following from RFC 3261:
"If the Request-URI indicates a resource at this proxy that does not
   exist, the proxy MUST return a 404 (Not Found) response.

   If the target set remains empty after applying all of the above, the
   proxy MUST return an error response, which SHOULD be the 480
   (Temporarily Unavailable) response."

Doesn't this fit into the first category, suggesting that 404 would be
appropriate? Or do we regard the existence of an AoR as meaning that
both SIP and SIPS resources exist, even if one or other has no
registered contacts?

John
 

> -----Original Message-----
> From: Francois Audet [mailto:audet@nortel.com] 
> Sent: 26 July 2007 22:03
> To: IETF SIP List
> Subject: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> 
> In the meeting, the issue of the error code to be used when the wrong
> URI is
> used in a request-URI (i.e., sip instead of sips, or vice-versa) came
> up.
> 
> The status quo is 2 new error codes (418 and 419). Another 
> proposal was
> to use only one
> (i.e., 418 with 2 new headers).
> 
> That proposal did not seem to get much traction. And people were not
> very with the
> status quo either. 
> 
> Another proposal was to use 480 (Temporarily Unavailable) and 
> NOT to use
> an explicit
> indication of the required/supported URIs (SIP, SIPS, etc.). 
> Of course,
> one can always
> put text in the reason phrase to describe the error for displaying to
> the end-use.
> 
> After talking to all the individual who seemed to have strong opinions
> on this topic,
> it appears that the 480 approach is the preferred approach. 
> 
> Please let the list know if you are strongly opposed to this outcome,
> along with the 
> reasons behind your opposition.
> 
> This is the last remaining issue to be closed in the draft, 
> as a result
> of WGLC.
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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From sip-bounces@ietf.org Thu Jul 26 18:42:06 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Thu, 26 Jul 2007 17:41:59 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Elwell, John" <john.elwell@siemens.com>, "IETF SIP List" <sip@ietf.org>
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=20

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]=20
> Sent: Thursday, July 26, 2007 15:38
> To: Audet, Francois (SC100:3055); IETF SIP List
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> Francois,
>=20
> I am reasonably happy with this proposal, BUT I must point=20
> out the following from RFC 3261:
> "If the Request-URI indicates a resource at this proxy that does not
>    exist, the proxy MUST return a 404 (Not Found) response.
>=20
>    If the target set remains empty after applying all of the=20
> above, the
>    proxy MUST return an error response, which SHOULD be the 480
>    (Temporarily Unavailable) response."
>=20
> Doesn't this fit into the first category, suggesting that 404=20
> would be appropriate? Or do we regard the existence of an AoR=20
> as meaning that both SIP and SIPS resources exist, even if=20
> one or other has no registered contacts?

Basically, 480 is the error code that is supposed to be used when
the user is known to exist (i.e., the AOR is in the database), but=20
he is not "logged-in" at present. So the presence of the AOR does
mean that both the SIP and SIPS AOR exist, but they don't necessarily
have a binding assigned to it at the time. Thus the 480.

That's the feedback I got from others.


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Date: Thu, 26 Jul 2007 17:59:32 -0500
To: "Francois Audet" <audet@nortel.com>, "IETF SIP List" <sip@ietf.org>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Francois

I just don't see the difference between a 480 and a 400 in this case. 
Either way, the UAC is left clueless, and hopeless (because it 
doesn't understand the real reason things went south).

I prefer telling the UAC what the error was. This is making a policy 
decision for the user of the UAC at the protocol level without their 
ability to make a choice, while blinding them to what actually happened.

I can give lots of examples outside of calling where this would be 
endlessly frustrating, and a few that are within calling...

At 05:21 PM 7/26/2007, Francois Audet wrote:
>Hi James,
>
>The specific reason why many were advocating a 480 instead of a new
>error code is exactly because they did NOT want the error code
>to encourage UAC to reattempt the request by downgrading to
>SIP automatically.
>
>This is the point that Eric Rescorla was making in the meeting, and
>it has been re-iterated in one way or another by both Jon Peterson
>and by Rohan Mahy.
>
>You *could* of course with this approach use a reason phrase in the
>480 to provide a message to the user of what the issue is.
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Thursday, July 26, 2007 14:31
> > To: Audet, Francois (SC100:3055); IETF SIP List
> > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> > At 04:03 PM 7/26/2007, Francois Audet wrote:
> > >In the meeting, the issue of the error code to be used when
> > the wrong
> > >URI is used in a request-URI (i.e., sip instead of sips, or
> > vice-versa) came up.
> > >
> > >The status quo is 2 new error codes (418 and 419).
> >
> > (BTW 419 is used by Cullen's hashcash draft)
> >
> > >Another proposal was to use only one (i.e., 418 with 2 new headers).
> >
> > I would advocate this proposal, especially verses a 480, as a
> > (new) Reason code to give the specifics can easily be
> > stripped along the way back to the UAC, leaving it clueless
> > as to what the actual problem was. Also not all
> > implementations support Reason (though they should).
> >
> > A 418 is specific to the point of what was wrong.... and all
> > UACs have to deal with the fact they can become a 400 (thus
> > removing the details)
> >
> >
> > >That proposal did not seem to get much traction. And people were not
> > >very with the status quo either.
> > >
> > >Another proposal was to use 480 (Temporarily Unavailable)
> > and NOT to use
> > >an explicit indication of the required/supported URIs (SIP, SIPS,
> > >etc.). Of course,
> > >one can always put text in the reason phrase to describe the error
> > >for displaying to
> > >the end-use.
> > >
> > >After talking to all the individual who seemed to have
> > strong opinions
> > >on this topic, it appears that the 480 approach is the
> > preferred approach.
> > >
> > >Please let the list know if you are strongly opposed to this outcome,
> > >along with the reasons behind your opposition.
> > >
> > >This is the last remaining issue to be closed in the draft,
> > as a result
> > >of WGLC.
> > >
> > >
> > >_______________________________________________
> > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >This list is for NEW development of the core SIP Protocol
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sipping@ietf.org for new developments on the application of sip
> >


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Robert Sparks <rjsparks@nostrum.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Is the 480 intended to be *indistinguishable* from a 480 returned as a 
result of no registrations at all? Or is it intended to be 
distinguishable but more obscure than an explicit error response?

I think it needs to be distinguishable, one way or another.

The reason is that there are UIs where it makes sense to downgrade. For 
instance:

The input to the UI of the UA is just a dial string, not a URI. It is up 
to the UA to decide what scheme to apply. It can do this based on 
policy, or perhaps based on a "secure" button on the phone. Typically 
the user won't know whether a particular number should be sip or sips. 
(Or even what that means.) The phone may decide to try sips first and 
downgrade to sip if the sips doesn't work. If sips works then it can 
display some sort of indication that the call is more secure than if sip 
is used.

If the UA can't tell why the sips request failed then it will have to 
retry the call in all cases.

The bottom line is that whether to downgrade is a valid policy issue, 
and we shouldn't arbitrarily block it from working.

	Paul

Robert Sparks wrote:
> I think the 480 approach described below is the right thing to do.
> 
> RjS
> 
> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
> 
>> In the meeting, the issue of the error code to be used when the wrong
>> URI is
>> used in a request-URI (i.e., sip instead of sips, or vice-versa) came
>> up.
>>
>> The status quo is 2 new error codes (418 and 419). Another proposal was
>> to use only one
>> (i.e., 418 with 2 new headers).
>>
>> That proposal did not seem to get much traction. And people were not
>> very with the
>> status quo either.
>>
>> Another proposal was to use 480 (Temporarily Unavailable) and NOT to use
>> an explicit
>> indication of the required/supported URIs (SIP, SIPS, etc.). Of course,
>> one can always
>> put text in the reason phrase to describe the error for displaying to
>> the end-use.
>>
>> After talking to all the individual who seemed to have strong opinions
>> on this topic,
>> it appears that the 480 approach is the preferred approach.
>>
>> Please let the list know if you are strongly opposed to this outcome,
>> along with the
>> reasons behind your opposition.
>>
>> This is the last remaining issue to be closed in the draft, as a result
>> of WGLC.
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Fri Jul 27 01:34:11 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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=20
>>The bottom line is that whether to downgrade is a valid
>>policy issue, and we shouldn't arbitrarily block it from working.

I would agree with this.

I think a UA user would always like the most secure call possible.
If the user only wants secure calls, then it can configure that on
the phone and so this would block any automatic downgrade.

In an other situation, the user could hear a warning tone or
message saying that the security downgrade is about occur.
Then if the user doesn't like it they can just hang up.

As Paul says, downgrade is valid.

Regards,
Attila




-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: 27 July 2007 00:17
To: Robert Sparks
Cc: IETF SIP List; Francois Audet
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418

Is the 480 intended to be *indistinguishable* from a 480 returned as a
result of no registrations at all? Or is it intended to be
distinguishable but more obscure than an explicit error response?

I think it needs to be distinguishable, one way or another.

The reason is that there are UIs where it makes sense to downgrade. For
instance:

The input to the UI of the UA is just a dial string, not a URI. It is up
to the UA to decide what scheme to apply. It can do this based on
policy, or perhaps based on a "secure" button on the phone. Typically
the user won't know whether a particular number should be sip or sips.=20
(Or even what that means.) The phone may decide to try sips first and
downgrade to sip if the sips doesn't work. If sips works then it can
display some sort of indication that the call is more secure than if sip
is used.

If the UA can't tell why the sips request failed then it will have to
retry the call in all cases.

The bottom line is that whether to downgrade is a valid policy issue,
and we shouldn't arbitrarily block it from working.

	Paul

Robert Sparks wrote:
> I think the 480 approach described below is the right thing to do.
>=20
> RjS
>=20
> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
>=20
>> In the meeting, the issue of the error code to be used when the wrong

>> URI is used in a request-URI (i.e., sip instead of sips, or=20
>> vice-versa) came up.
>>
>> The status quo is 2 new error codes (418 and 419). Another proposal=20
>> was to use only one (i.e., 418 with 2 new headers).
>>
>> That proposal did not seem to get much traction. And people were not=20
>> very with the status quo either.
>>
>> Another proposal was to use 480 (Temporarily Unavailable) and NOT to=20
>> use an explicit indication of the required/supported URIs (SIP, SIPS,

>> etc.). Of course, one can always put text in the reason phrase to=20
>> describe the error for displaying to the end-use.
>>
>> After talking to all the individual who seemed to have strong=20
>> opinions on this topic, it appears that the 480 approach is the=20
>> preferred approach.
>>
>> Please let the list know if you are strongly opposed to this outcome,

>> along with the reasons behind your opposition.
>>
>> This is the last remaining issue to be closed in the draft, as a=20
>> result of WGLC.
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=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


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


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



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To: "'Paul Kyzivat'" <pkyzivat@cisco.com>,
	"'Robert Sparks'" <rjsparks@nostrum.com>
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	<46A92B7C.7080202@cisco.com>
Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 01:59:07 -0400
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I agree with Paul mostly. 
Especially since I think we would also want the UAC to *upgrade* if it tried
sip but the AoR was only registered at sips. (note I mean the UAC upgrading,
not the proxy of course)
Obfuscating the error seems wrong. (clever though :)

-hadriel

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, July 26, 2007 7:17 PM
> To: Robert Sparks
> Cc: IETF SIP List; Francois Audet
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> 
> Is the 480 intended to be *indistinguishable* from a 480 returned as a
> result of no registrations at all? Or is it intended to be
> distinguishable but more obscure than an explicit error response?
> 
> I think it needs to be distinguishable, one way or another.
> 
> The reason is that there are UIs where it makes sense to downgrade. For
> instance:
> 
> The input to the UI of the UA is just a dial string, not a URI. It is up
> to the UA to decide what scheme to apply. It can do this based on
> policy, or perhaps based on a "secure" button on the phone. Typically
> the user won't know whether a particular number should be sip or sips.
> (Or even what that means.) The phone may decide to try sips first and
> downgrade to sip if the sips doesn't work. If sips works then it can
> display some sort of indication that the call is more secure than if sip
> is used.
> 
> If the UA can't tell why the sips request failed then it will have to
> retry the call in all cases.
> 
> The bottom line is that whether to downgrade is a valid policy issue,
> and we shouldn't arbitrarily block it from working.
> 
> 	Paul
> 
> Robert Sparks wrote:
> > I think the 480 approach described below is the right thing to do.
> >
> > RjS
> >
> > On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
> >
> >> In the meeting, the issue of the error code to be used when the wrong
> >> URI is
> >> used in a request-URI (i.e., sip instead of sips, or vice-versa) came
> >> up.
> >>
> >> The status quo is 2 new error codes (418 and 419). Another proposal was
> >> to use only one
> >> (i.e., 418 with 2 new headers).
> >>
> >> That proposal did not seem to get much traction. And people were not
> >> very with the
> >> status quo either.
> >>
> >> Another proposal was to use 480 (Temporarily Unavailable) and NOT to
> use
> >> an explicit
> >> indication of the required/supported URIs (SIP, SIPS, etc.). Of course,
> >> one can always
> >> put text in the reason phrase to describe the error for displaying to
> >> the end-use.
> >>
> >> After talking to all the individual who seemed to have strong opinions
> >> on this topic,
> >> it appears that the 480 approach is the preferred approach.
> >>
> >> Please let the list know if you are strongly opposed to this outcome,
> >> along with the
> >> reasons behind your opposition.
> >>
> >> This is the last remaining issue to be closed in the draft, as a result
> >> of WGLC.
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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



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Date: Fri, 27 Jul 2007 17:08:15 +1000
From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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I also agree with Paul's views. Can someone elaborate as to why it is not
desirable for a UAC to retry automatically? This could be a check box in the
setting menu.

Regards,
Hisham


On 27/07/07, Paul Kyzivat <pkyzivat@cisco.com> wrote:
>
> Is the 480 intended to be *indistinguishable* from a 480 returned as a
> result of no registrations at all? Or is it intended to be
> distinguishable but more obscure than an explicit error response?
>
> I think it needs to be distinguishable, one way or another.
>
> The reason is that there are UIs where it makes sense to downgrade. For
> instance:
>
> The input to the UI of the UA is just a dial string, not a URI. It is up
> to the UA to decide what scheme to apply. It can do this based on
> policy, or perhaps based on a "secure" button on the phone. Typically
> the user won't know whether a particular number should be sip or sips.
> (Or even what that means.) The phone may decide to try sips first and
> downgrade to sip if the sips doesn't work. If sips works then it can
> display some sort of indication that the call is more secure than if sip
> is used.
>
> If the UA can't tell why the sips request failed then it will have to
> retry the call in all cases.
>
> The bottom line is that whether to downgrade is a valid policy issue,
> and we shouldn't arbitrarily block it from working.
>
>        Paul
>
> Robert Sparks wrote:
> > I think the 480 approach described below is the right thing to do.
> >
> > RjS
> >
> > On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
> >
> >> In the meeting, the issue of the error code to be used when the wrong
> >> URI is
> >> used in a request-URI (i.e., sip instead of sips, or vice-versa) came
> >> up.
> >>
> >> The status quo is 2 new error codes (418 and 419). Another proposal was
> >> to use only one
> >> (i.e., 418 with 2 new headers).
> >>
> >> That proposal did not seem to get much traction. And people were not
> >> very with the
> >> status quo either.
> >>
> >> Another proposal was to use 480 (Temporarily Unavailable) and NOT to
> use
> >> an explicit
> >> indication of the required/supported URIs (SIP, SIPS, etc.). Of course,
> >> one can always
> >> put text in the reason phrase to describe the error for displaying to
> >> the end-use.
> >>
> >> After talking to all the individual who seemed to have strong opinions
> >> on this topic,
> >> it appears that the 480 approach is the preferred approach.
> >>
> >> Please let the list know if you are strongly opposed to this outcome,
> >> along with the
> >> reasons behind your opposition.
> >>
> >> This is the last remaining issue to be closed in the draft, as a result
> >> of WGLC.
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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<div>I also agree with Paul&#39;s views. Can someone elaborate as to why it is not desirable for a UAC to retry automatically? This could be a check box in the setting menu.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>Hisham<br><br>&nbsp;</div>
<div><span class="gmail_quote">On 27/07/07, <b class="gmail_sendername">Paul Kyzivat</b> &lt;<a href="mailto:pkyzivat@cisco.com">pkyzivat@cisco.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Is the 480 intended to be *indistinguishable* from a 480 returned as a<br>result of no registrations at all? Or is it intended to be
<br>distinguishable but more obscure than an explicit error response?<br><br>I think it needs to be distinguishable, one way or another.<br><br>The reason is that there are UIs where it makes sense to downgrade. For<br>instance:
<br><br>The input to the UI of the UA is just a dial string, not a URI. It is up<br>to the UA to decide what scheme to apply. It can do this based on<br>policy, or perhaps based on a &quot;secure&quot; button on the phone. Typically
<br>the user won&#39;t know whether a particular number should be sip or sips.<br>(Or even what that means.) The phone may decide to try sips first and<br>downgrade to sip if the sips doesn&#39;t work. If sips works then it can
<br>display some sort of indication that the call is more secure than if sip<br>is used.<br><br>If the UA can&#39;t tell why the sips request failed then it will have to<br>retry the call in all cases.<br><br>The bottom line is that whether to downgrade is a valid policy issue,
<br>and we shouldn&#39;t arbitrarily block it from working.<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br><br>Robert Sparks wrote:<br>&gt; I think the 480 approach described below is the right thing to do.<br>&gt;<br>&gt; RjS<br>&gt;<br>&gt; On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
<br>&gt;<br>&gt;&gt; In the meeting, the issue of the error code to be used when the wrong<br>&gt;&gt; URI is<br>&gt;&gt; used in a request-URI (i.e., sip instead of sips, or vice-versa) came<br>&gt;&gt; up.<br>&gt;&gt;<br>
&gt;&gt; The status quo is 2 new error codes (418 and 419). Another proposal was<br>&gt;&gt; to use only one<br>&gt;&gt; (i.e., 418 with 2 new headers).<br>&gt;&gt;<br>&gt;&gt; That proposal did not seem to get much traction. And people were not
<br>&gt;&gt; very with the<br>&gt;&gt; status quo either.<br>&gt;&gt;<br>&gt;&gt; Another proposal was to use 480 (Temporarily Unavailable) and NOT to use<br>&gt;&gt; an explicit<br>&gt;&gt; indication of the required/supported URIs (SIP, SIPS, etc.). Of course,
<br>&gt;&gt; one can always<br>&gt;&gt; put text in the reason phrase to describe the error for displaying to<br>&gt;&gt; the end-use.<br>&gt;&gt;<br>&gt;&gt; After talking to all the individual who seemed to have strong opinions
<br>&gt;&gt; on this topic,<br>&gt;&gt; it appears that the 480 approach is the preferred approach.<br>&gt;&gt;<br>&gt;&gt; Please let the list know if you are strongly opposed to this outcome,<br>&gt;&gt; along with the<br>
&gt;&gt; reasons behind your opposition.<br>&gt;&gt;<br>&gt;&gt; This is the last remaining issue to be closed in the draft, as a result<br>&gt;&gt; of WGLC.<br>&gt;&gt;<br>&gt;&gt;<br>&gt;&gt; _______________________________________________
<br>&gt;&gt; Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt;&gt; This list is for NEW development of the core SIP Protocol<br>&gt;&gt; Use <a href="mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip<br>&gt;&gt; Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br>&gt;<br>&gt;<br>&gt;<br>&gt; _______________________________________________
<br>&gt; Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt; This list is for NEW development of the core SIP Protocol<br>&gt; Use <a href="mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip<br>&gt; Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br>&gt;<br><br><br>_______________________________________________
<br>Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a href="mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip<br>Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br></blockquote></div><br>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1269492937==--





From sip-bounces@ietf.org Fri Jul 27 10:35:42 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 09:35:12 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <Jon.Peterson@neustar.com>,
	"Rohan Mahy" <rohan@ekabal.com>, "Eric Rescorla" <ekr@networkresonance.com>
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The intent is to be UNDISTINGUISHABLE.

We do NOT want the equipment to automatically downgrade. We want
the user to make the decision in a concious way.

Furthermore, we want to emphasise that the SIP and SIPS are=20
different addresses and are not interchangeable.

Rohan Mahy, Jon Peterson, Eric Rescorla,

Since you were the main people advocating this change, can=20
you make clear on the list what the rationale is.

Thanks.
=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Thursday, July 26, 2007 16:17
> To: Robert Sparks
> Cc: Audet, Francois (SC100:3055); IETF SIP List
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> Is the 480 intended to be *indistinguishable* from a 480=20
> returned as a result of no registrations at all? Or is it=20
> intended to be distinguishable but more obscure than an=20
> explicit error response?
>=20
> I think it needs to be distinguishable, one way or another.
>=20
> The reason is that there are UIs where it makes sense to=20
> downgrade. For
> instance:
>=20
> The input to the UI of the UA is just a dial string, not a=20
> URI. It is up to the UA to decide what scheme to apply. It=20
> can do this based on policy, or perhaps based on a "secure"=20
> button on the phone. Typically the user won't know whether a=20
> particular number should be sip or sips.=20
> (Or even what that means.) The phone may decide to try sips=20
> first and downgrade to sip if the sips doesn't work. If sips=20
> works then it can display some sort of indication that the=20
> call is more secure than if sip is used.
>=20
> If the UA can't tell why the sips request failed then it will=20
> have to retry the call in all cases.
>=20
> The bottom line is that whether to downgrade is a valid=20
> policy issue, and we shouldn't arbitrarily block it from working.
>=20
> 	Paul
>=20
> Robert Sparks wrote:
> > I think the 480 approach described below is the right thing to do.
> >=20
> > RjS
> >=20
> > On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
> >=20
> >> In the meeting, the issue of the error code to be used=20
> when the wrong=20
> >> URI is used in a request-URI (i.e., sip instead of sips, or=20
> >> vice-versa) came up.
> >>
> >> The status quo is 2 new error codes (418 and 419). Another=20
> proposal=20
> >> was to use only one (i.e., 418 with 2 new headers).
> >>
> >> That proposal did not seem to get much traction. And=20
> people were not=20
> >> very with the status quo either.
> >>
> >> Another proposal was to use 480 (Temporarily Unavailable)=20
> and NOT to=20
> >> use an explicit indication of the required/supported URIs=20
> (SIP, SIPS,=20
> >> etc.). Of course, one can always put text in the reason phrase to=20
> >> describe the error for displaying to the end-use.
> >>
> >> After talking to all the individual who seemed to have strong=20
> >> opinions on this topic, it appears that the 480 approach is the=20
> >> preferred approach.
> >>
> >> Please let the list know if you are strongly opposed to=20
> this outcome,=20
> >> along with the reasons behind your opposition.
> >>
> >> This is the last remaining issue to be closed in the draft, as a=20
> >> result of WGLC.
> >>
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use=20
> >> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >> sipping@ietf.org for new developments on the application of sip
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20


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From sip-bounces@ietf.org Fri Jul 27 10:40:25 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 09:40:11 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "James M. Polk" <jmpolk@cisco.com>, "IETF SIP List" <sip@ietf.org>
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400 =3D=3D Bad Request (i.e., there is something wrong with it)

480 =3D=3D Temporarily Unavailable (i.e., there is currently no
       Contact binding associated with the URI)=20

If I send a request to a SIPS Request-URI and there is no
current binding to a SIPS Contact, then I will receive a
480.

If I send a where the Request-URI is SIPS, but the Contact
is SIP, then I get a 400.

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Thursday, July 26, 2007 16:00
> To: Audet, Francois (SC100:3055); IETF SIP List
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> Francois
>=20
> I just don't see the difference between a 480 and a 400 in this case.=20
> Either way, the UAC is left clueless, and hopeless (because=20
> it doesn't understand the real reason things went south).
>=20
> I prefer telling the UAC what the error was. This is making a=20
> policy decision for the user of the UAC at the protocol level=20
> without their ability to make a choice, while blinding them=20
> to what actually happened.
>=20
> I can give lots of examples outside of calling where this=20
> would be endlessly frustrating, and a few that are within calling...
>=20
> At 05:21 PM 7/26/2007, Francois Audet wrote:
> >Hi James,
> >
> >The specific reason why many were advocating a 480 instead of a new=20
> >error code is exactly because they did NOT want the error code to=20
> >encourage UAC to reattempt the request by downgrading to SIP=20
> >automatically.
> >
> >This is the point that Eric Rescorla was making in the=20
> meeting, and it=20
> >has been re-iterated in one way or another by both Jon=20
> Peterson and by=20
> >Rohan Mahy.
> >
> >You *could* of course with this approach use a reason phrase=20
> in the 480=20
> >to provide a message to the user of what the issue is.
> >
> > > -----Original Message-----
> > > From: James M. Polk [mailto:jmpolk@cisco.com]
> > > Sent: Thursday, July 26, 2007 14:31
> > > To: Audet, Francois (SC100:3055); IETF SIP List
> > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > > At 04:03 PM 7/26/2007, Francois Audet wrote:
> > > >In the meeting, the issue of the error code to be used when
> > > the wrong
> > > >URI is used in a request-URI (i.e., sip instead of sips, or
> > > vice-versa) came up.
> > > >
> > > >The status quo is 2 new error codes (418 and 419).
> > >
> > > (BTW 419 is used by Cullen's hashcash draft)
> > >
> > > >Another proposal was to use only one (i.e., 418 with 2=20
> new headers).
> > >
> > > I would advocate this proposal, especially verses a 480, as a
> > > (new) Reason code to give the specifics can easily be=20
> stripped along=20
> > > the way back to the UAC, leaving it clueless as to what=20
> the actual=20
> > > problem was. Also not all implementations support Reason (though=20
> > > they should).
> > >
> > > A 418 is specific to the point of what was wrong.... and all UACs=20
> > > have to deal with the fact they can become a 400 (thus=20
> removing the=20
> > > details)
> > >
> > >
> > > >That proposal did not seem to get much traction. And people were=20
> > > >not very with the status quo either.
> > > >
> > > >Another proposal was to use 480 (Temporarily Unavailable)
> > > and NOT to use
> > > >an explicit indication of the required/supported URIs=20
> (SIP, SIPS,=20
> > > >etc.). Of course, one can always put text in the reason=20
> phrase to=20
> > > >describe the error for displaying to the end-use.
> > > >
> > > >After talking to all the individual who seemed to have
> > > strong opinions
> > > >on this topic, it appears that the 480 approach is the
> > > preferred approach.
> > > >
> > > >Please let the list know if you are strongly opposed to this=20
> > > >outcome, along with the reasons behind your opposition.
> > > >
> > > >This is the last remaining issue to be closed in the draft,
> > > as a result
> > > >of WGLC.
> > > >
> > > >
> > > >_______________________________________________
> > > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >This list is for NEW development of the core SIP Protocol Use=20
> > > >sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > >sipping@ietf.org for new developments on the application of sip
> > >
>=20


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To: Francois Audet <audet@nortel.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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You mean, security through obscurity? I guess it's about par for
SIPS course.

       Mike

Francois Audet wrote:
> The intent is to be UNDISTINGUISHABLE.
>
> We do NOT want the equipment to automatically downgrade. We want
> the user to make the decision in a concious way.
>
> Furthermore, we want to emphasise that the SIP and SIPS are 
> different addresses and are not interchangeable.
>
> Rohan Mahy, Jon Peterson, Eric Rescorla,
>
> Since you were the main people advocating this change, can 
> you make clear on the list what the rationale is.
>
> Thanks.
>  
>
>   
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Thursday, July 26, 2007 16:17
>> To: Robert Sparks
>> Cc: Audet, Francois (SC100:3055); IETF SIP List
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> Is the 480 intended to be *indistinguishable* from a 480 
>> returned as a result of no registrations at all? Or is it 
>> intended to be distinguishable but more obscure than an 
>> explicit error response?
>>
>> I think it needs to be distinguishable, one way or another.
>>
>> The reason is that there are UIs where it makes sense to 
>> downgrade. For
>> instance:
>>
>> The input to the UI of the UA is just a dial string, not a 
>> URI. It is up to the UA to decide what scheme to apply. It 
>> can do this based on policy, or perhaps based on a "secure" 
>> button on the phone. Typically the user won't know whether a 
>> particular number should be sip or sips. 
>> (Or even what that means.) The phone may decide to try sips 
>> first and downgrade to sip if the sips doesn't work. If sips 
>> works then it can display some sort of indication that the 
>> call is more secure than if sip is used.
>>
>> If the UA can't tell why the sips request failed then it will 
>> have to retry the call in all cases.
>>
>> The bottom line is that whether to downgrade is a valid 
>> policy issue, and we shouldn't arbitrarily block it from working.
>>
>> 	Paul
>>
>> Robert Sparks wrote:
>>     
>>> I think the 480 approach described below is the right thing to do.
>>>
>>> RjS
>>>
>>> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
>>>
>>>       
>>>> In the meeting, the issue of the error code to be used 
>>>>         
>> when the wrong 
>>     
>>>> URI is used in a request-URI (i.e., sip instead of sips, or 
>>>> vice-versa) came up.
>>>>
>>>> The status quo is 2 new error codes (418 and 419). Another 
>>>>         
>> proposal 
>>     
>>>> was to use only one (i.e., 418 with 2 new headers).
>>>>
>>>> That proposal did not seem to get much traction. And 
>>>>         
>> people were not 
>>     
>>>> very with the status quo either.
>>>>
>>>> Another proposal was to use 480 (Temporarily Unavailable) 
>>>>         
>> and NOT to 
>>     
>>>> use an explicit indication of the required/supported URIs 
>>>>         
>> (SIP, SIPS, 
>>     
>>>> etc.). Of course, one can always put text in the reason phrase to 
>>>> describe the error for displaying to the end-use.
>>>>
>>>> After talking to all the individual who seemed to have strong 
>>>> opinions on this topic, it appears that the 480 approach is the 
>>>> preferred approach.
>>>>
>>>> Please let the list know if you are strongly opposed to 
>>>>         
>> this outcome, 
>>     
>>>> along with the reasons behind your opposition.
>>>>
>>>> This is the last remaining issue to be closed in the draft, as a 
>>>> result of WGLC.
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use 
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>> sipping@ietf.org for new developments on the application of sip
>>>>         
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>>       
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>   


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 10:06:18 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Michael Thomas" <mat@cisco.com>
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Oh, please with the rethoric.=20

Again, the idea is that a sip address is different than a sips address.

If you use a sip address when a sips is expected, then you'd get=20
the behavior you'd get for when there is no binging assigned.

It's as simple as that.

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]=20
> Sent: Friday, July 27, 2007 07:58
> To: Audet, Francois (SC100:3055)
> Cc: Paul Kyzivat; Jon.Peterson@neustar.com; Rohan Mahy; Eric=20
> Rescorla; IETF SIP List
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> You mean, security through obscurity? I guess it's about par=20
> for SIPS course.
>=20
>        Mike
>=20
> Francois Audet wrote:
> > The intent is to be UNDISTINGUISHABLE.
> >
> > We do NOT want the equipment to automatically downgrade. We=20
> want the=20
> > user to make the decision in a concious way.
> >
> > Furthermore, we want to emphasise that the SIP and SIPS are=20
> different=20
> > addresses and are not interchangeable.
> >
> > Rohan Mahy, Jon Peterson, Eric Rescorla,
> >
> > Since you were the main people advocating this change, can you make=20
> > clear on the list what the rationale is.
> >
> > Thanks.
> > =20
> >
> >  =20
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Thursday, July 26, 2007 16:17
> >> To: Robert Sparks
> >> Cc: Audet, Francois (SC100:3055); IETF SIP List
> >> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >>
> >> Is the 480 intended to be *indistinguishable* from a 480=20
> returned as=20
> >> a result of no registrations at all? Or is it intended to be=20
> >> distinguishable but more obscure than an explicit error response?
> >>
> >> I think it needs to be distinguishable, one way or another.
> >>
> >> The reason is that there are UIs where it makes sense to=20
> downgrade.=20
> >> For
> >> instance:
> >>
> >> The input to the UI of the UA is just a dial string, not a=20
> URI. It is=20
> >> up to the UA to decide what scheme to apply. It can do=20
> this based on=20
> >> policy, or perhaps based on a "secure"
> >> button on the phone. Typically the user won't know whether a=20
> >> particular number should be sip or sips.
> >> (Or even what that means.) The phone may decide to try=20
> sips first and=20
> >> downgrade to sip if the sips doesn't work. If sips works=20
> then it can=20
> >> display some sort of indication that the call is more=20
> secure than if=20
> >> sip is used.
> >>
> >> If the UA can't tell why the sips request failed then it=20
> will have to=20
> >> retry the call in all cases.
> >>
> >> The bottom line is that whether to downgrade is a valid=20
> policy issue,=20
> >> and we shouldn't arbitrarily block it from working.
> >>
> >> 	Paul
> >>
> >> Robert Sparks wrote:
> >>    =20
> >>> I think the 480 approach described below is the right thing to do.
> >>>
> >>> RjS
> >>>
> >>> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
> >>>
> >>>      =20
> >>>> In the meeting, the issue of the error code to be used
> >>>>        =20
> >> when the wrong
> >>    =20
> >>>> URI is used in a request-URI (i.e., sip instead of sips, or
> >>>> vice-versa) came up.
> >>>>
> >>>> The status quo is 2 new error codes (418 and 419). Another
> >>>>        =20
> >> proposal
> >>    =20
> >>>> was to use only one (i.e., 418 with 2 new headers).
> >>>>
> >>>> That proposal did not seem to get much traction. And
> >>>>        =20
> >> people were not
> >>    =20
> >>>> very with the status quo either.
> >>>>
> >>>> Another proposal was to use 480 (Temporarily Unavailable)
> >>>>        =20
> >> and NOT to
> >>    =20
> >>>> use an explicit indication of the required/supported URIs
> >>>>        =20
> >> (SIP, SIPS,
> >>    =20
> >>>> etc.). Of course, one can always put text in the reason=20
> phrase to=20
> >>>> describe the error for displaying to the end-use.
> >>>>
> >>>> After talking to all the individual who seemed to have strong=20
> >>>> opinions on this topic, it appears that the 480 approach is the=20
> >>>> preferred approach.
> >>>>
> >>>> Please let the list know if you are strongly opposed to
> >>>>        =20
> >> this outcome,
> >>    =20
> >>>> along with the reasons behind your opposition.
> >>>>
> >>>> This is the last remaining issue to be closed in the draft, as a=20
> >>>> result of WGLC.
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >>>>        =20
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=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
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >  =20
>=20


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



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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Francois Audet <audet@nortel.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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	<247B95AF-63F6-432D-9D6E-AF083AF9C525@nostrum.com>
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I understand that if the user entered a sips URI then it should be the 
user that must decide to downgrade. But if the user didn't know whether 
to use sip or sips in the first place, and the UA decides to try sips 
first then I see no problem in the UA having a policy that causes it to 
downgrade.

This is highly related to how security is managed in the user interface.

	Paul

Francois Audet wrote:
> The intent is to be UNDISTINGUISHABLE.
> 
> We do NOT want the equipment to automatically downgrade. We want
> the user to make the decision in a concious way.
> 
> Furthermore, we want to emphasise that the SIP and SIPS are 
> different addresses and are not interchangeable.
> 
> Rohan Mahy, Jon Peterson, Eric Rescorla,
> 
> Since you were the main people advocating this change, can 
> you make clear on the list what the rationale is.
> 
> Thanks.
>  
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Thursday, July 26, 2007 16:17
>> To: Robert Sparks
>> Cc: Audet, Francois (SC100:3055); IETF SIP List
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> Is the 480 intended to be *indistinguishable* from a 480 
>> returned as a result of no registrations at all? Or is it 
>> intended to be distinguishable but more obscure than an 
>> explicit error response?
>>
>> I think it needs to be distinguishable, one way or another.
>>
>> The reason is that there are UIs where it makes sense to 
>> downgrade. For
>> instance:
>>
>> The input to the UI of the UA is just a dial string, not a 
>> URI. It is up to the UA to decide what scheme to apply. It 
>> can do this based on policy, or perhaps based on a "secure" 
>> button on the phone. Typically the user won't know whether a 
>> particular number should be sip or sips. 
>> (Or even what that means.) The phone may decide to try sips 
>> first and downgrade to sip if the sips doesn't work. If sips 
>> works then it can display some sort of indication that the 
>> call is more secure than if sip is used.
>>
>> If the UA can't tell why the sips request failed then it will 
>> have to retry the call in all cases.
>>
>> The bottom line is that whether to downgrade is a valid 
>> policy issue, and we shouldn't arbitrarily block it from working.
>>
>> 	Paul
>>
>> Robert Sparks wrote:
>>> I think the 480 approach described below is the right thing to do.
>>>
>>> RjS
>>>
>>> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
>>>
>>>> In the meeting, the issue of the error code to be used 
>> when the wrong 
>>>> URI is used in a request-URI (i.e., sip instead of sips, or 
>>>> vice-versa) came up.
>>>>
>>>> The status quo is 2 new error codes (418 and 419). Another 
>> proposal 
>>>> was to use only one (i.e., 418 with 2 new headers).
>>>>
>>>> That proposal did not seem to get much traction. And 
>> people were not 
>>>> very with the status quo either.
>>>>
>>>> Another proposal was to use 480 (Temporarily Unavailable) 
>> and NOT to 
>>>> use an explicit indication of the required/supported URIs 
>> (SIP, SIPS, 
>>>> etc.). Of course, one can always put text in the reason phrase to 
>>>> describe the error for displaying to the end-use.
>>>>
>>>> After talking to all the individual who seemed to have strong 
>>>> opinions on this topic, it appears that the 480 approach is the 
>>>> preferred approach.
>>>>
>>>> Please let the list know if you are strongly opposed to 
>> this outcome, 
>>>> along with the reasons behind your opposition.
>>>>
>>>> This is the last remaining issue to be closed in the draft, as a 
>>>> result of WGLC.
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use 
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>> sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>
> 


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



From sip-bounces@ietf.org Fri Jul 27 11:20:58 2007
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 10:20:31 -0500
To: "Francois Audet" <audet@nortel.com>
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Hi,

I think 480 has the correct semantics.  When an authoritative proxy  
has no Contacts registered for a resource (in this case it is a  
"sips" resource), the correct response code is 480.  In addition we  
don't want the UAC to automatically downgrade because it mistakenly  
thinks that's the right thing to do.

My intent however was not to make this indistinguishable, since I  
proposed adding a Warning: header.  I think  it is useful to have the  
reason there where no relevant Contacts.

thanks,
-rohan



On Jul 27, 2007, at 9:35 AM, Francois Audet wrote:

> The intent is to be UNDISTINGUISHABLE.
>
> We do NOT want the equipment to automatically downgrade. We want
> the user to make the decision in a concious way.
>
> Furthermore, we want to emphasise that the SIP and SIPS are
> different addresses and are not interchangeable.
>
> Rohan Mahy, Jon Peterson, Eric Rescorla,
>
> Since you were the main people advocating this change, can
> you make clear on the list what the rationale is.
>
> Thanks.
>
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Thursday, July 26, 2007 16:17
>> To: Robert Sparks
>> Cc: Audet, Francois (SC100:3055); IETF SIP List
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> Is the 480 intended to be *indistinguishable* from a 480
>> returned as a result of no registrations at all? Or is it
>> intended to be distinguishable but more obscure than an
>> explicit error response?
>>
>> I think it needs to be distinguishable, one way or another.
>>
>> The reason is that there are UIs where it makes sense to
>> downgrade. For
>> instance:
>>
>> The input to the UI of the UA is just a dial string, not a
>> URI. It is up to the UA to decide what scheme to apply. It
>> can do this based on policy, or perhaps based on a "secure"
>> button on the phone. Typically the user won't know whether a
>> particular number should be sip or sips.
>> (Or even what that means.) The phone may decide to try sips
>> first and downgrade to sip if the sips doesn't work. If sips
>> works then it can display some sort of indication that the
>> call is more secure than if sip is used.
>>
>> If the UA can't tell why the sips request failed then it will
>> have to retry the call in all cases.
>>
>> The bottom line is that whether to downgrade is a valid
>> policy issue, and we shouldn't arbitrarily block it from working.
>>
>> 	Paul
>>
>> Robert Sparks wrote:
>>> I think the 480 approach described below is the right thing to do.
>>>
>>> RjS
>>>
>>> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
>>>
>>>> In the meeting, the issue of the error code to be used
>> when the wrong
>>>> URI is used in a request-URI (i.e., sip instead of sips, or
>>>> vice-versa) came up.
>>>>
>>>> The status quo is 2 new error codes (418 and 419). Another
>> proposal
>>>> was to use only one (i.e., 418 with 2 new headers).
>>>>
>>>> That proposal did not seem to get much traction. And
>> people were not
>>>> very with the status quo either.
>>>>
>>>> Another proposal was to use 480 (Temporarily Unavailable)
>> and NOT to
>>>> use an explicit indication of the required/supported URIs
>> (SIP, SIPS,
>>>> etc.). Of course, one can always put text in the reason phrase to
>>>> describe the error for displaying to the end-use.
>>>>
>>>> After talking to all the individual who seemed to have strong
>>>> opinions on this topic, it appears that the 480 approach is the
>>>> preferred approach.
>>>>
>>>> Please let the list know if you are strongly opposed to
>> this outcome,
>>>> along with the reasons behind your opposition.
>>>>
>>>> This is the last remaining issue to be closed in the draft, as a
>>>> result of WGLC.
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>> sipping@ietf.org for new developments on the application of sip
>>>
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use
>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>



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



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From: Michael Thomas <mat@cisco.com>
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To: Francois Audet <audet@nortel.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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I don't see the point in parsimony on return codes unless you're trying to
obscure something. And I don't find "We don't want [...]" particularly
compelling as it implies there's a one-size-fits-all on UI design. Were
SIPS an end to end mechanism there _might_ be a case to be made (and
I'd be extremely dubious even then), but it's far more broken than that
so making proscriptions against this or that is rather pointless.

Never try to teach a *pig* to sing; it wastes your time and it *annoys 
the pig*. *...* Mark *Twain

       Mike
*
Francois Audet wrote:
> Oh, please with the rethoric. 
>
> Again, the idea is that a sip address is different than a sips address.
>
> If you use a sip address when a sips is expected, then you'd get 
> the behavior you'd get for when there is no binging assigned.
>
> It's as simple as that.
>
>   
>> -----Original Message-----
>> From: Michael Thomas [mailto:mat@cisco.com] 
>> Sent: Friday, July 27, 2007 07:58
>> To: Audet, Francois (SC100:3055)
>> Cc: Paul Kyzivat; Jon.Peterson@neustar.com; Rohan Mahy; Eric 
>> Rescorla; IETF SIP List
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> You mean, security through obscurity? I guess it's about par 
>> for SIPS course.
>>
>>        Mike
>>
>> Francois Audet wrote:
>>     
>>> The intent is to be UNDISTINGUISHABLE.
>>>
>>> We do NOT want the equipment to automatically downgrade. We 
>>>       
>> want the 
>>     
>>> user to make the decision in a concious way.
>>>
>>> Furthermore, we want to emphasise that the SIP and SIPS are 
>>>       
>> different 
>>     
>>> addresses and are not interchangeable.
>>>
>>> Rohan Mahy, Jon Peterson, Eric Rescorla,
>>>
>>> Since you were the main people advocating this change, can you make 
>>> clear on the list what the rationale is.
>>>
>>> Thanks.
>>>  
>>>
>>>   
>>>       
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Sent: Thursday, July 26, 2007 16:17
>>>> To: Robert Sparks
>>>> Cc: Audet, Francois (SC100:3055); IETF SIP List
>>>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>>>
>>>> Is the 480 intended to be *indistinguishable* from a 480 
>>>>         
>> returned as 
>>     
>>>> a result of no registrations at all? Or is it intended to be 
>>>> distinguishable but more obscure than an explicit error response?
>>>>
>>>> I think it needs to be distinguishable, one way or another.
>>>>
>>>> The reason is that there are UIs where it makes sense to 
>>>>         
>> downgrade. 
>>     
>>>> For
>>>> instance:
>>>>
>>>> The input to the UI of the UA is just a dial string, not a 
>>>>         
>> URI. It is 
>>     
>>>> up to the UA to decide what scheme to apply. It can do 
>>>>         
>> this based on 
>>     
>>>> policy, or perhaps based on a "secure"
>>>> button on the phone. Typically the user won't know whether a 
>>>> particular number should be sip or sips.
>>>> (Or even what that means.) The phone may decide to try 
>>>>         
>> sips first and 
>>     
>>>> downgrade to sip if the sips doesn't work. If sips works 
>>>>         
>> then it can 
>>     
>>>> display some sort of indication that the call is more 
>>>>         
>> secure than if 
>>     
>>>> sip is used.
>>>>
>>>> If the UA can't tell why the sips request failed then it 
>>>>         
>> will have to 
>>     
>>>> retry the call in all cases.
>>>>
>>>> The bottom line is that whether to downgrade is a valid 
>>>>         
>> policy issue, 
>>     
>>>> and we shouldn't arbitrarily block it from working.
>>>>
>>>> 	Paul
>>>>
>>>> Robert Sparks wrote:
>>>>     
>>>>         
>>>>> I think the 480 approach described below is the right thing to do.
>>>>>
>>>>> RjS
>>>>>
>>>>> On Jul 26, 2007, at 4:03 PM, Francois Audet wrote:
>>>>>
>>>>>       
>>>>>           
>>>>>> In the meeting, the issue of the error code to be used
>>>>>>         
>>>>>>             
>>>> when the wrong
>>>>     
>>>>         
>>>>>> URI is used in a request-URI (i.e., sip instead of sips, or
>>>>>> vice-versa) came up.
>>>>>>
>>>>>> The status quo is 2 new error codes (418 and 419). Another
>>>>>>         
>>>>>>             
>>>> proposal
>>>>     
>>>>         
>>>>>> was to use only one (i.e., 418 with 2 new headers).
>>>>>>
>>>>>> That proposal did not seem to get much traction. And
>>>>>>         
>>>>>>             
>>>> people were not
>>>>     
>>>>         
>>>>>> very with the status quo either.
>>>>>>
>>>>>> Another proposal was to use 480 (Temporarily Unavailable)
>>>>>>         
>>>>>>             
>>>> and NOT to
>>>>     
>>>>         
>>>>>> use an explicit indication of the required/supported URIs
>>>>>>         
>>>>>>             
>>>> (SIP, SIPS,
>>>>     
>>>>         
>>>>>> etc.). Of course, one can always put text in the reason 
>>>>>>             
>> phrase to 
>>     
>>>>>> describe the error for displaying to the end-use.
>>>>>>
>>>>>> After talking to all the individual who seemed to have strong 
>>>>>> opinions on this topic, it appears that the 480 approach is the 
>>>>>> preferred approach.
>>>>>>
>>>>>> Please let the list know if you are strongly opposed to
>>>>>>         
>>>>>>             
>>>> this outcome,
>>>>     
>>>>         
>>>>>> along with the reasons behind your opposition.
>>>>>>
>>>>>> This is the last remaining issue to be closed in the draft, as a 
>>>>>> result of WGLC.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol Use 
>>>>>> sip-implementors@cs.columbia.edu for questions on 
>>>>>>             
>> current sip Use 
>>     
>>>>>> sipping@ietf.org for new developments on the application of sip
>>>>>>         
>>>>>>             
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol Use 
>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>>>> sipping@ietf.org for new developments on the application of sip
>>>>>
>>>>>       
>>>>>           
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sipping@ietf.org for new developments on the application of sip
>>>   
>>>       


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From sip-bounces@ietf.org Fri Jul 27 11:33:04 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 10:32:48 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Alan Hawrylyshen" <alan@polyphase.ca>,
	"Kyzivat Paul" <pkyzivat@cisco.com>, "Mahy Rohan" <rohan@ekabal.com>
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On the Warning header...

Values 300-379 are reserved for SDP-related stuff.

Values 380-389 are unassiged (but there is no text on it).

Values 390-398 are unassigned (but don't fall into the SDP
related stuff.=20

399 means Miscellaneous and the text can be provided to the user.
It further says that Automata MUST NOT take action based on this.

Rohan, are you advocating using 399, or defining a new code from the
390-398 space?

> -----Original Message-----
> From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com] On=20
> Behalf Of Alan Hawrylyshen
> Sent: Friday, July 27, 2007 08:27
> To: Kyzivat Paul; Mahy Rohan
> Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> Jon.Peterson@neustar.com
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
>=20
> On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
>=20
> > I understand that if the user entered a sips URI then it=20
> should be the=20
> > user that must decide to downgrade. But if the user didn't know=20
> > whether to use sip or sips in the first place, and the UA=20
> decides to=20
> > try sips first then I see no problem in the UA having a policy that=20
> > causes it to downgrade.
>=20
> I was under the impression (based on meeting discussion) that :
> 1 - the downgrade was undesirable because it reveals=20
> (possibly) information about the targeted party in the clear, and;
> 2 - The 480 with a Warning header was an option to provide=20
> automata- friendly indication of the failure reason.
>=20
> Alan Hawrylyshen
>=20
> a l a n a t p o l y p h a s e d o t c a
>=20
>=20
>=20


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From: Eric Rescorla <ekr@networkresonance.com>
To: "Francois Audet" <audet@nortel.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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At Fri, 27 Jul 2007 09:35:12 -0500,
Francois Audet wrote:
> The intent is to be UNDISTINGUISHABLE.
> 
> We do NOT want the equipment to automatically downgrade. We want
> the user to make the decision in a concious way.
> 
> Furthermore, we want to emphasise that the SIP and SIPS are 
> different addresses and are not interchangeable.
> 
> Rohan Mahy, Jon Peterson, Eric Rescorla,
> 
> Since you were the main people advocating this change, can 
> you make clear on the list what the rationale is.

Right. So, I want to avoid automatic downgrade. I think that
at minimum that means that we should avoid using a return
code ordinary associated with automatic retry. Better yet
would be to use a hard failure return code and then a textual
error message...

-Ekr


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Eric Rescorla wrote:
> At Fri, 27 Jul 2007 09:35:12 -0500,
> Francois Audet wrote:
>   
>> The intent is to be UNDISTINGUISHABLE.
>>
>> We do NOT want the equipment to automatically downgrade. We want
>> the user to make the decision in a concious way.
>>
>> Furthermore, we want to emphasise that the SIP and SIPS are 
>> different addresses and are not interchangeable.
>>
>> Rohan Mahy, Jon Peterson, Eric Rescorla,
>>
>> Since you were the main people advocating this change, can 
>> you make clear on the list what the rationale is.
>>     
>
> Right. So, I want to avoid automatic downgrade. I think that
> at minimum that means that we should avoid using a return
> code ordinary associated with automatic retry. Better yet
> would be to use a hard failure return code and then a textual
> error message...
>   

Do people seriously believe that any amount of protestation from the IETF
is going to prevent implementors from downgrading? If the receiver(s) really
don't want the SIP method, they don't have to accept them. If they do, 
then no
amount of hand wringing MUST NOT's is going to prevent the inevitable.

The only thing this kind of hand wringing is likely to produce is more
incompatibility between the puritans and the sinners.

       Mike


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From sip-bounces@ietf.org Fri Jul 27 12:02:31 2007
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Fri, 27 Jul 2007 11:02:12 -0500
To: Michael Thomas <mat@cisco.com>
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Michael,

At issue here is what the default implementor is likely to do.  With  
a new 4xx, the misguided but well-meaning implementor is likely to  
try to "helpfully" "repair" the error without thinking about or  
understanding the security context.

Using a Warning code raises the bar significantly, but still allows  
automata to at least log what happened.

thanks,
-rohan



On Jul 27, 2007, at 10:47 AM, Michael Thomas wrote:

> Eric Rescorla wrote:
>> At Fri, 27 Jul 2007 09:35:12 -0500,
>> Francois Audet wrote:
>>
>>> The intent is to be UNDISTINGUISHABLE.
>>>
>>> We do NOT want the equipment to automatically downgrade. We want
>>> the user to make the decision in a concious way.
>>>
>>> Furthermore, we want to emphasise that the SIP and SIPS are  
>>> different addresses and are not interchangeable.
>>>
>>> Rohan Mahy, Jon Peterson, Eric Rescorla,
>>>
>>> Since you were the main people advocating this change, can you  
>>> make clear on the list what the rationale is.
>>>
>>
>> Right. So, I want to avoid automatic downgrade. I think that
>> at minimum that means that we should avoid using a return
>> code ordinary associated with automatic retry. Better yet
>> would be to use a hard failure return code and then a textual
>> error message...
>>
>
> Do people seriously believe that any amount of protestation from  
> the IETF
> is going to prevent implementors from downgrading? If the receiver 
> (s) really
> don't want the SIP method, they don't have to accept them. If they  
> do, then no
> amount of hand wringing MUST NOT's is going to prevent the inevitable.
>
> The only thing this kind of hand wringing is likely to produce is more
> incompatibility between the puritans and the sinners.
>
>       Mike



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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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=20
> Do people seriously believe that any amount of protestation=20
> from the IETF is going to prevent implementors from=20
> downgrading? If the receiver(s) really don't want the SIP=20
> method, they don't have to accept them. If they do, then no=20
> amount of hand wringing MUST NOT's is going to prevent the inevitable.
>=20
> The only thing this kind of hand wringing is likely to=20
> produce is more incompatibility between the puritans and the sinners.

Doesn't this argue agaist your own previous email?

The 480 approach doesn't allow for automatic retry, because you
can't figure out from the error code that it might actually work=20
by downgrading.

I'm am quite lost at understanding what concrete outcome you want
to see.


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Rohan Mahy wrote:
> Michael,
>
> At issue here is what the default implementor is likely to do.  With a 
> new 4xx, the misguided but well-meaning implementor is likely to try 
> to "helpfully" "repair" the error without thinking about or 
> understanding the security context.
>
> Using a Warning code raises the bar significantly, but still allows 
> automata to at least log what happened.
As I said, a receiver is completely at liberty to prevent the downgrade 
by not
accepting the downgraded request.  Problem solved by those who care. On
the other hand I think it's pretty presumptuous that we "know" in all cases
whether it's wrong to "repair" the error. Paul brought up several examples.
Also: if the receiver allows non-SIPS requests why shouldn't you infer that
they want you to "repair" the request? It sure seems like this is a 
misguided
attempt at forcing ineffectual nannyware which is a pointless enterprise for
IETF protocols.

       Mike


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A 399 Miscellaneous?  That doesn't make sense...

Warning codes are not limited to 3XX, so if a new Warning code is 
what Rohan wants to convey more information in the response (which I 
believe should be done one way or another) - it can be almost any 
value other than 300-379 or 399 IMO.

At 10:32 AM 7/27/2007, Francois Audet wrote:
>On the Warning header...
>
>Values 300-379 are reserved for SDP-related stuff.
>
>Values 380-389 are unassiged (but there is no text on it).
>
>Values 390-398 are unassigned (but don't fall into the SDP
>related stuff.
>
>399 means Miscellaneous and the text can be provided to the user.
>It further says that Automata MUST NOT take action based on this.
>
>Rohan, are you advocating using 399, or defining a new code from the
>390-398 space?
>
> > -----Original Message-----
> > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com] On
> > Behalf Of Alan Hawrylyshen
> > Sent: Friday, July 27, 2007 08:27
> > To: Kyzivat Paul; Mahy Rohan
> > Cc: Audet, Francois (SC100:3055); IETF SIP List;
> > Jon.Peterson@neustar.com
> > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> >
> > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> >
> > > I understand that if the user entered a sips URI then it
> > should be the
> > > user that must decide to downgrade. But if the user didn't know
> > > whether to use sip or sips in the first place, and the UA
> > decides to
> > > try sips first then I see no problem in the UA having a policy that
> > > causes it to downgrade.
> >
> > I was under the impression (based on meeting discussion) that :
> > 1 - the downgrade was undesirable because it reveals
> > (possibly) information about the targeted party in the clear, and;
> > 2 - The 480 with a Warning header was an option to provide
> > automata- friendly indication of the failure reason.
> >
> > Alan Hawrylyshen
> >
> > a l a n a t p o l y p h a s e d o t c a
> >
> >
> >
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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Alan Hawrylyshen wrote:
> 
> On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> 
>> I understand that if the user entered a sips URI then it should be the 
>> user that must decide to downgrade. But if the user didn't know 
>> whether to use sip or sips in the first place, and the UA decides to 
>> try sips first then I see no problem in the UA having a policy that 
>> causes it to downgrade.
> 
> I was under the impression (based on meeting discussion) that :
> 1 - the downgrade was undesirable because it reveals (possibly) 
> information about the targeted party in the clear, and;

Its only undesirable if the UAC had any expectation that the information 
would not be revealed. I repeat that this is largely a function of the 
UI and contract between the UI and the user. I don't think we can or 
should make assumptions about that that is.

> 2 - The 480 with a Warning header was an option to provide 
> automata-friendly indication of the failure reason.

I don't have strong feelings about what the response is, so long as the 
particular case is distinguishable in some way.

	Thanks,
	Paul

> Alan Hawrylyshen
> 
> a l a n a t p o l y p h a s e d o t c a
> 


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From sip-bounces@ietf.org Mon Jul 30 04:28:17 2007
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From: ji ma <mjhorse1984@yahoo.com.cn>
Subject: [Sip]RFC4916:the usage of connected identity in interdomain
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Hi folks:
   
      I have some questions after reading RFC4916, which descibes a solution of ensuring the identity of the UA that sends backward request after a dialog has formed. As far as I'm concerned, this specification is a supplement of RFC4474, which can only provide the validaty of the UA sending the request. However, RFC4916's 2 examples regarding to the its desciption only illustrate how to convey connected identity in an assumption that the caller and callee are in the same domain (example.com in particular), by using UPDATE messages which renew the From field of the sender. So my question is, can this specification resolve the authentication of the connected identity regarding to the scenories where the caller and callee are in different domains?
  [Considering RFC4474, the authenticator will not be the responsible proxy for the request that contains a From field (bob@biloxi.example.com) that is different from its real identity (bob@chicago.example.com) ]
   
      Another doubt I'm concerning about, is the content of last two paragraphs in page 3 of RFC4916, which is said as below:
   
     The provision of the identity of the responder in a response
     (commonly called "response identity") is outside the scope of this
     document.
   
        Note that even if identity were to be conveyed somehow in a
        response, there would in general be difficulty authenticating the
        UAS.  Providing identity in a separate request allows normal
        authentication techniques to be used.
   
   
  I can't understand why it's difficult to authenticate response messages as I thought it could be authenticated in a way like the request it responses to(I don't know how much different in the process it should be regarding request and response) .So what's special for response?
   
  Regards,
  Ma Ji
      

       
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<div>Hi folks:</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp; I have some questions after reading RFC4916, which descibes a solution of ensuring the identity of the UA that sends backward request after a dialog has formed. As far as I'm concerned, this specification is a supplement of RFC4474, which can only provide the validaty of the UA sending the request. However, RFC4916's 2 examples regarding to the its desciption only&nbsp;illustrate how to convey connected identity&nbsp;in an assumption that the caller and callee are&nbsp;in the&nbsp;same domain (example.com in particular), by using UPDATE messages which renew the From field of the sender. So my question is, can this specification resolve the authentication of the&nbsp;connected&nbsp;identity regarding to the scenories where the caller and callee are in different domains?</div>  <div>[Considering RFC4474, the authenticator will not be the responsible proxy for the request that contains a From field (<SPAN
 lang=EN-US style="FONT-SIZE: 10.5pt; FONT-FAMILY: 'Times New Roman'; mso-bidi-font-size: 12.0pt; mso-fareast-font-family: ËÎÌå; mso-font-kerning: 1.0pt; mso-ansi-language: EN-US; mso-fareast-language: ZH-CN; mso-bidi-language: AR-SA"><A href="mailto:bob@biloxi.example.com">bob@biloxi.example.com</A></SPAN>) that is different from its real identity (<SPAN lang=EN-US style="FONT-SIZE: 10.5pt; FONT-FAMILY: 'Times New Roman'; mso-bidi-font-size: 12.0pt; mso-fareast-font-family: ËÎÌå; mso-font-kerning: 1.0pt; mso-ansi-language: EN-US; mso-fareast-language: ZH-CN; mso-bidi-language: AR-SA"><A href="mailto:bob@chicago.example.com">bob@chicago.example.com</A></SPAN>) ]</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp; Another doubt I'm concerning about, is the content of last two paragraphs in page 3 of RFC4916, which is said as below:</div>  <div>&nbsp;</div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><A style="mso-comment-reference: mj_1; mso-comment-date:
 20070627T0949"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp; </SPAN>The provision of the identity of the responder in a response</FONT></FONT></SPAN></A></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN style="mso-comment-continuation: 1"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp; </SPAN>(commonly called "response identity") is outside the scope of this</FONT></FONT></SPAN></SPAN></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN style="mso-comment-continuation: 1"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp; </SPAN>document.</FONT></FONT></SPAN></SPAN></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN class=MsoCommentReference><SPAN lang=EN-US><SPAN style="mso-special-character: comment"><FONT face="Times New Roman"
 size=3></FONT></SPAN></SPAN></SPAN>&nbsp;</div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>Note that even if identity were to be conveyed somehow in a</FONT></FONT></SPAN></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>response, there would in general be difficulty authenticating the</FONT></FONT></SPAN></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3><FONT face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>UAS.<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>Providing identity in a separate request allows normal</FONT></FONT></SPAN></div>  <div class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3><FONT
 face="Times New Roman"><SPAN style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>authentication techniques to be used.</FONT></FONT></SPAN></div>  <DIV style="mso-element: comment-list"><FONT face="Times New Roman" size=3></FONT>&nbsp;</DIV>  <div>&nbsp;</div>  <div>I can't understand why it's difficult to authenticate response messages as I thought it could be authenticated in a way like the request it responses to(I don't know how much different in the process it should be regarding request and response) .So what's special for response?</div>  <div>&nbsp;</div>  <div>Regards,</div>  <div>Ma Ji</div>  <div>&nbsp;&nbsp;&nbsp; </div><p>&#32;



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Ma Ji,

See below.

John

> -----Original Message-----
> From: ji ma [mailto:mjhorse1984@yahoo.com.cn]=20
> Sent: 30 July 2007 09:28
> To: sip@ietf.org
> Subject: [Sip]RFC4916:the usage of connected identity in interdomain
>=20
> Hi folks:
> =20
>     I have some questions after reading RFC4916, which=20
> descibes a solution of ensuring the identity of the UA that=20
> sends backward request after a dialog has formed. As far as=20
> I'm concerned, this specification is a supplement of RFC4474,=20
> which can only provide the validaty of the UA sending the=20
> request. However, RFC4916's 2 examples regarding to the its=20
> desciption only illustrate how to convey connected identity=20
> in an assumption that the caller and callee are in the same=20
> domain (example.com in particular), by using UPDATE messages=20
> which renew the From field of the sender. So my question is,=20
> can this specification resolve the authentication of the=20
> connected identity regarding to the scenories where the=20
> caller and callee are in different domains?
> [Considering RFC4474, the authenticator will not be the=20
> responsible proxy for the request that contains a From field=20
> (bob@biloxi.example.com) that is different from its real=20
> identity (bob@chicago.example.com) ]
[JRE] Although the examples show only a single domain, the RFC is =
equally applicable when caller and callee are in different domains. Each =
domain will provide its own authentication service.

> =20
>     Another doubt I'm concerning about, is the content of=20
> last two paragraphs in page 3 of RFC4916, which is said as below:
> =20
>    The provision of the identity of the responder in a response
>    (commonly called "response identity") is outside the scope of this
>    document.
> =20
>       Note that even if identity were to be conveyed somehow in a
>       response, there would in general be difficulty=20
> authenticating the
>       UAS.  Providing identity in a separate request allows normal
>       authentication techniques to be used.
> =20
> =20
> I can't understand why it's difficult to authenticate=20
> response messages as I thought it could be authenticated in a=20
> way like the request it responses to(I don't know how much=20
> different in the process it should be regarding request and=20
> response) .So what's special for response?
[JRE] SIP digest authentication is only specified for requests. In =
particular, there is no mechanism for challenging a response.

> =20
> Regards,
> Ma Ji
>    =20
>=20
> ________________________________
>=20
> =
=C7=C0=D7=A2=D1=C5=BB=A2=C3=E2=B7=D1=D3=CA=CF=E43.5G=C8=DD=C1=BF=A3=AC20M=
=B8=BD=BC=FE=A3=A1 <http://cn.mail.yahoo.com> =20
>=20


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



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Date: Mon, 30 Jul 2007 19:15:27 +0800 (CST)
From: ji ma <mjhorse1984@yahoo.com.cn>
Subject: RE: [Sip]RFC4916:the usage of connected identity in interdomain
To: "Elwell, John" <john.elwell@siemens.com>, sip@ietf.org
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"Elwell, John" <john.elwell@siemens.com> Ð´µÀ£º  Ma Ji,

See below.

John

> -----Original Message-----
> From: ji ma [mailto:mjhorse1984@yahoo.com.cn] 
> Sent: 30 July 2007 09:28
> To: sip@ietf.org
> Subject: [Sip]RFC4916:the usage of connected identity in interdomain
> 
> Hi folks:
> 
> I have some questions after reading RFC4916, which 
> descibes a solution of ensuring the identity of the UA that 
> sends backward request after a dialog has formed. As far as 
> I'm concerned, this specification is a supplement of RFC4474, 
> which can only provide the validaty of the UA sending the 
> request. However, RFC4916's 2 examples regarding to the its 
> desciption only illustrate how to convey connected identity 
> in an assumption that the caller and callee are in the same 
> domain (example.com in particular), by using UPDATE messages 
> which renew the From field of the sender. So my question is, 
> can this specification resolve the authentication of the 
> connected identity regarding to the scenories where the 
> caller and callee are in different domains?
> [Considering RFC4474, the authenticator will not be the 
> responsible proxy for the request that contains a From field 
> (bob@biloxi.example.com) that is different from its real 
> identity (bob@chicago.example.com) ]
[JRE] Although the examples show only a single domain, the RFC is equally applicable when caller and callee are in different domains. Each domain will provide its own authentication service.

> 
> Another doubt I'm concerning about, is the content of 
> last two paragraphs in page 3 of RFC4916, which is said as below:
> 
> The provision of the identity of the responder in a response
> (commonly called "response identity") is outside the scope of this
> document.
> 
> Note that even if identity were to be conveyed somehow in a
> response, there would in general be difficulty 
> authenticating the
> UAS. Providing identity in a separate request allows normal
> authentication techniques to be used.
> 
> 
> I can't understand why it's difficult to authenticate 
> response messages as I thought it could be authenticated in a 
> way like the request it responses to(I don't know how much 
> different in the process it should be regarding request and 
> response) .So what's special for response?
[JRE] SIP digest authentication is only specified for requests. In particular, there is no mechanism for challenging a response.

> 
> Regards,
> Ma Ji
  Thanks for reminding. When using connected identity,the sender has no reason to use the previous identity the same as the From URI of the request, who sends UPDATE which contains the new identity of the sender.
   
   More, I deliberate that authenticating the validaty of the receiver in a response(that means the To header) is also required, and so far I has not see any draft nor maillist discussing it(maybe I missed), considering the current mechanism is not enough for such issue, I thing there should be some effort to be put to take this aspect into account.
   
  Regards,
  Ma Ji

       
---------------------------------
ÇÀ×¢ÑÅ»¢Ãâ·ÑÓÊÏä3.5GÈÝÁ¿£¬20M¸½¼þ£¡ 
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<BR><BR><B><I>"Elwell, John" &lt;john.elwell@siemens.com&gt;</I></B> Ð´µÀ£º  <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Ma Ji,<BR><BR>See below.<BR><BR>John<BR><BR>&gt; -----Original Message-----<BR>&gt; From: ji ma [mailto:mjhorse1984@yahoo.com.cn] <BR>&gt; Sent: 30 July 2007 09:28<BR>&gt; To: sip@ietf.org<BR>&gt; Subject: [Sip]RFC4916:the usage of connected identity in interdomain<BR>&gt; <BR>&gt; Hi folks:<BR>&gt; <BR>&gt; I have some questions after reading RFC4916, which <BR>&gt; descibes a solution of ensuring the identity of the UA that <BR>&gt; sends backward request after a dialog has formed. As far as <BR>&gt; I'm concerned, this specification is a supplement of RFC4474, <BR>&gt; which can only provide the validaty of the UA sending the <BR>&gt; request. However, RFC4916's 2 examples regarding to the its <BR>&gt; desciption only illustrate how to convey connected identity <BR>&gt; in an assumption that the
 caller and callee are in the same <BR>&gt; domain (example.com in particular), by using UPDATE messages <BR>&gt; which renew the From field of the sender. So my question is, <BR>&gt; can this specification resolve the authentication of the <BR>&gt; connected identity regarding to the scenories where the <BR>&gt; caller and callee are in different domains?<BR>&gt; [Considering RFC4474, the authenticator will not be the <BR>&gt; responsible proxy for the request that contains a From field <BR>&gt; (bob@biloxi.example.com) that is different from its real <BR>&gt; identity (bob@chicago.example.com) ]<BR>[JRE] Although the examples show only a single domain, the RFC is equally applicable when caller and callee are in different domains. Each domain will provide its own authentication service.<BR><BR>&gt; <BR>&gt; Another doubt I'm concerning about, is the content of <BR>&gt; last two paragraphs in page 3 of RFC4916, which is said as below:<BR>&gt; <BR>&gt; The provision of the
 identity of the responder in a response<BR>&gt; (commonly called "response identity") is outside the scope of this<BR>&gt; document.<BR>&gt; <BR>&gt; Note that even if identity were to be conveyed somehow in a<BR>&gt; response, there would in general be difficulty <BR>&gt; authenticating the<BR>&gt; UAS. Providing identity in a separate request allows normal<BR>&gt; authentication techniques to be used.<BR>&gt; <BR>&gt; <BR>&gt; I can't understand why it's difficult to authenticate <BR>&gt; response messages as I thought it could be authenticated in a <BR>&gt; way like the request it responses to(I don't know how much <BR>&gt; different in the process it should be regarding request and <BR>&gt; response) .So what's special for response?<BR>[JRE] SIP digest authentication is only specified for requests. In particular, there is no mechanism for challenging a response.<BR><BR>&gt; <BR>&gt; Regards,<BR>&gt; Ma Ji<BR></BLOCKQUOTE>  <div>Thanks for reminding. When using
 connected identity,the sender has no reason to use the previous identity&nbsp;the same as&nbsp;the From URI of the request,&nbsp;who sends&nbsp;UPDATE which contains the new identity of the sender.</div>  <div>&nbsp;</div>  <div>&nbsp;More, I&nbsp;deliberate that&nbsp;authenticating the validaty of the receiver in a response(that means the To header) is also required, and so far I has not see any draft nor maillist discussing it(maybe I missed),&nbsp;considering the current mechanism is not enough for such issue, I thing there should be some effort to be put to take this aspect into account.</div>  <div>&nbsp;</div>  <div>Regards,</div>  <div>Ma Ji</div><p>&#32;



      <hr size=1><a href="http://cn.mail.yahoo.com" target=blank>ÇÀ×¢ÑÅ»¢Ãâ·ÑÓÊÏä3.5GÈÝÁ¿£¬20M¸½¼þ£¡</a> 
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Date: Mon, 30 Jul 2007 14:01:30 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip]RFC4916:the usage of connected identity in interdomain
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>  More, I deliberate that authenticating the validaty of the=20
> receiver in a response(that means the To header) is also=20
> required, and so far I has not see any draft nor maillist=20
> discussing it(maybe I missed), considering the current=20
> mechanism is not enough for such issue, I thing there should=20
> be some effort to be put to take this aspect into account.
[JRE] There have been a number of proposals in the past for response =
identity, but the problems encountered have proved to be insurmountable. =
I would suggest you look in the SIP and SIPPING mail archives and =
minutes from late 2005 / early 2006. Note also that, because of request =
retargeting, the identity of the responder will not necessarily equate =
to the identity in the To header field. In the absence of any solution =
to the response identity problem, RFC 4916 fills the gap by allowing =
authenticated identity to occur in a request in the reverse direction.

John
> =20
> Regards,
> Ma Ji
>=20
> ________________________________
>=20
> =
=C7=C0=D7=A2=D1=C5=BB=A2=C3=E2=B7=D1=D3=CA=CF=E43.5G=C8=DD=C1=BF=A3=AC20M=
=B8=BD=BC=FE=A3=A1 <http://cn.mail.yahoo.com> =20
>=20


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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RFC 3261, p. 181:

	"The first digit  of warning codes beginning with "3" indicates
warnings
	specific to SIP."

Then it explains within the 3XX space, how they are used.

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Saturday, July 28, 2007 18:25
> To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat=20
> Paul; Mahy Rohan
> Cc: IETF SIP List; Jon.Peterson@neustar.com
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> A 399 Miscellaneous?  That doesn't make sense...
>=20
> Warning codes are not limited to 3XX, so if a new Warning=20
> code is what Rohan wants to convey more information in the=20
> response (which I believe should be done one way or another)=20
> - it can be almost any value other than 300-379 or 399 IMO.
>=20
> At 10:32 AM 7/27/2007, Francois Audet wrote:
> >On the Warning header...
> >
> >Values 300-379 are reserved for SDP-related stuff.
> >
> >Values 380-389 are unassiged (but there is no text on it).
> >
> >Values 390-398 are unassigned (but don't fall into the SDP related=20
> >stuff.
> >
> >399 means Miscellaneous and the text can be provided to the user.
> >It further says that Automata MUST NOT take action based on this.
> >
> >Rohan, are you advocating using 399, or defining a new code from the
> >390-398 space?
> >
> > > -----Original Message-----
> > > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com]=20
> On Behalf Of=20
> > > Alan Hawrylyshen
> > > Sent: Friday, July 27, 2007 08:27
> > > To: Kyzivat Paul; Mahy Rohan
> > > Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> > > Jon.Peterson@neustar.com
> > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > >
> > > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> > >
> > > > I understand that if the user entered a sips URI then it
> > > should be the
> > > > user that must decide to downgrade. But if the user didn't know=20
> > > > whether to use sip or sips in the first place, and the UA
> > > decides to
> > > > try sips first then I see no problem in the UA having a policy=20
> > > > that causes it to downgrade.
> > >
> > > I was under the impression (based on meeting discussion) that :
> > > 1 - the downgrade was undesirable because it reveals
> > > (possibly) information about the targeted party in the clear, and;
> > > 2 - The 480 with a Warning header was an option to provide
> > > automata- friendly indication of the failure reason.
> > >
> > > Alan Hawrylyshen
> > >
> > > a l a n a t p o l y p h a s e d o t c a
> > >
> > >
> > >
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol Use=20
> >sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 10:55:09 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Alan Hawrylyshen" <alan@polyphase.ca>,
	"Kyzivat Paul" <pkyzivat@cisco.com>, "Mahy Rohan" <rohan@ekabal.com>
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Correct.=20

> -----Original Message-----
> From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com] On=20
> Behalf Of Alan Hawrylyshen
> Sent: Friday, July 27, 2007 08:27
> To: Kyzivat Paul; Mahy Rohan
> Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> Jon.Peterson@neustar.com
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
>=20
> On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
>=20
> > I understand that if the user entered a sips URI then it=20
> should be the=20
> > user that must decide to downgrade. But if the user didn't know=20
> > whether to use sip or sips in the first place, and the UA=20
> decides to=20
> > try sips first then I see no problem in the UA having a policy that=20
> > causes it to downgrade.
>=20
> I was under the impression (based on meeting discussion) that :
> 1 - the downgrade was undesirable because it reveals=20
> (possibly) information about the targeted party in the clear, and;
> 2 - The 480 with a Warning header was an option to provide=20
> automata- friendly indication of the failure reason.
>=20
> Alan Hawrylyshen
>=20
> a l a n a t p o l y p h a s e d o t c a
>=20
>=20
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 10:54:57 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "James M. Polk" <jmpolk@cisco.com>, "Alan Hawrylyshen" <alan@polyphase.ca>,
	"Kyzivat Paul" <pkyzivat@cisco.com>, "Mahy Rohan" <rohan@ekabal.com>
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RFC 3261, p. 181:

	"The first digit  of warning codes beginning with "3" indicates
warnings
	specific to SIP."

Then it explains within the 3XX space, how they are used.

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Saturday, July 28, 2007 18:25
> To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat=20
> Paul; Mahy Rohan
> Cc: IETF SIP List; Jon.Peterson@neustar.com
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> A 399 Miscellaneous?  That doesn't make sense...
>=20
> Warning codes are not limited to 3XX, so if a new Warning=20
> code is what Rohan wants to convey more information in the=20
> response (which I believe should be done one way or another)=20
> - it can be almost any value other than 300-379 or 399 IMO.
>=20
> At 10:32 AM 7/27/2007, Francois Audet wrote:
> >On the Warning header...
> >
> >Values 300-379 are reserved for SDP-related stuff.
> >
> >Values 380-389 are unassiged (but there is no text on it).
> >
> >Values 390-398 are unassigned (but don't fall into the SDP related=20
> >stuff.
> >
> >399 means Miscellaneous and the text can be provided to the user.
> >It further says that Automata MUST NOT take action based on this.
> >
> >Rohan, are you advocating using 399, or defining a new code from the
> >390-398 space?
> >
> > > -----Original Message-----
> > > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com]=20
> On Behalf Of=20
> > > Alan Hawrylyshen
> > > Sent: Friday, July 27, 2007 08:27
> > > To: Kyzivat Paul; Mahy Rohan
> > > Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> > > Jon.Peterson@neustar.com
> > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > >
> > > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> > >
> > > > I understand that if the user entered a sips URI then it
> > > should be the
> > > > user that must decide to downgrade. But if the user didn't know=20
> > > > whether to use sip or sips in the first place, and the UA
> > > decides to
> > > > try sips first then I see no problem in the UA having a policy=20
> > > > that causes it to downgrade.
> > >
> > > I was under the impression (based on meeting discussion) that :
> > > 1 - the downgrade was undesirable because it reveals
> > > (possibly) information about the targeted party in the clear, and;
> > > 2 - The 480 with a Warning header was an option to provide
> > > automata- friendly indication of the failure reason.
> > >
> > > Alan Hawrylyshen
> > >
> > > a l a n a t p o l y p h a s e d o t c a
> > >
> > >
> > >
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol Use=20
> >sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 10:55:09 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Alan Hawrylyshen" <alan@polyphase.ca>,
	"Kyzivat Paul" <pkyzivat@cisco.com>, "Mahy Rohan" <rohan@ekabal.com>
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Correct.=20

> -----Original Message-----
> From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com] On=20
> Behalf Of Alan Hawrylyshen
> Sent: Friday, July 27, 2007 08:27
> To: Kyzivat Paul; Mahy Rohan
> Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> Jon.Peterson@neustar.com
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
>=20
> On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
>=20
> > I understand that if the user entered a sips URI then it=20
> should be the=20
> > user that must decide to downgrade. But if the user didn't know=20
> > whether to use sip or sips in the first place, and the UA=20
> decides to=20
> > try sips first then I see no problem in the UA having a policy that=20
> > causes it to downgrade.
>=20
> I was under the impression (based on meeting discussion) that :
> 1 - the downgrade was undesirable because it reveals=20
> (possibly) information about the targeted party in the clear, and;
> 2 - The 480 with a Warning header was an option to provide=20
> automata- friendly indication of the failure reason.
>=20
> Alan Hawrylyshen
>=20
> a l a n a t p o l y p h a s e d o t c a
>=20
>=20
>=20


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





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Date: Mon, 30 Jul 2007 09:08:44 -0700
From: Eric Rescorla <ekr@networkresonance.com>
To: Michael Thomas <mat@cisco.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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At Fri, 27 Jul 2007 09:39:06 -0700,
Michael Thomas wrote:
> 
> Rohan Mahy wrote:
> > Michael,
> >
> > At issue here is what the default implementor is likely to do.  With a 
> > new 4xx, the misguided but well-meaning implementor is likely to try 
> > to "helpfully" "repair" the error without thinking about or 
> > understanding the security context.
> >
> > Using a Warning code raises the bar significantly, but still allows 
> > automata to at least log what happened.
> As I said, a receiver is completely at liberty to prevent the downgrade 
> by not
> accepting the downgraded request.

Unless, of course, someone is impersonating the receiver.


>  Problem solved by those who care.

Regrettably no.

-Ekr


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Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Eric Rescorla wrote:
> At Fri, 27 Jul 2007 09:39:06 -0700,
> Michael Thomas wrote:
>   
>> Rohan Mahy wrote:
>>     
>>> Michael,
>>>
>>> At issue here is what the default implementor is likely to do.  With a 
>>> new 4xx, the misguided but well-meaning implementor is likely to try 
>>> to "helpfully" "repair" the error without thinking about or 
>>> understanding the security context.
>>>
>>> Using a Warning code raises the bar significantly, but still allows 
>>> automata to at least log what happened.
>>>       
>> As I said, a receiver is completely at liberty to prevent the downgrade 
>> by not
>> accepting the downgraded request.
>>     
>
> Unless, of course, someone is impersonating the receiver.
>   
Given how tangled up SIPS is, I really no idea what you're talking
about, or whether it's even responsive to my suggestion. Last I heard,
the entire raison d'etre of SIPS was that the next hop is cryptographically
identified via TLS. I'm guessing that you're not suggesting that TLS
is useless against impersonation attacks.


       Mike


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From sip-bounces@ietf.org Mon Jul 30 12:38:55 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 11:38:45 -0500
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From: "Francois Audet" <audet@nortel.com>
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In the interest of moving forward on this arcane issue of marginal
interest, I am making the
following concrete proposal.

Here are 2 alternatives. I'd like to have a virtual show of hands on
which option is=20
preferred by people.


OPTION 1

	Use Response code 480, with new Warning header field 390 (URI
Scheme Not Supported)
	for both cases when SIP or SIPS is rejected.

	The proxy or UAS may use the warning text field to provide more
indications, e.g.:

	Warning: 390 isi.edu "SIPS Not Allowed"
	Warning: 390 isi.edu "SIPS Required"

OPTION 2

	Use Response code 480, with 2 new Warning headera field 390
(SIPS Not Allowed)
	and 391 (SIPS Required).

	The proxy or UAS may use the warning text field to provide user,
e.g.:

	Warning: 390 isi.edu "Please use a SIP URI"
	Warning: 391 isi.edu "A secure SIPS URI is required"


OPTION 1 has the characteristic of being "opaque" to Automatas, but it
is generic
enough to support other URI schemes. The precise reasons is described in
human-readable
format.

OPTON 2 has the characteristic of precisely describing the reason for
the rejection. It
is NOT opaque to Automatas. If new warnings are necessary for other URI
shemes, new warning
headers can be defined.

In both cases, existing UACs (or UACs that do not wish to have any
special behavior based
on this) will exhibit the desired behavior, i.e., it will look like
"temporarily unavailable".


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From sip-bounces@ietf.org Mon Jul 30 13:13:56 2007
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Francois Audet wrote:
> In the interest of moving forward on this arcane issue of marginal
> interest, I am making the
> following concrete proposal.
. . .


> OPTION 2
> 
> 	Use Response code 480, with 2 new Warning headera field 390
> (SIPS Not Allowed)
> 	and 391 (SIPS Required).
> 
> 	The proxy or UAS may use the warning text field to provide user,
> e.g.:
> 
> 	Warning: 390 isi.edu "Please use a SIP URI"
> 	Warning: 391 isi.edu "A secure SIPS URI is required"
> 
> 

In the interest of coming to a conclusion, please sign me up for option 
2, but I'm willing to accept option 1 nearly as readily.


--
Dean


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I am ok with either of these, though I somewhat prefer Option 2.

I don't know what you mean about Option 1 being opaque to automata. It 
seems to me it provides comparable info to Option 2, giving that the UAC 
knows what it used in the request.

	Paul

Francois Audet wrote:
> In the interest of moving forward on this arcane issue of marginal
> interest, I am making the
> following concrete proposal.
> 
> Here are 2 alternatives. I'd like to have a virtual show of hands on
> which option is 
> preferred by people.
> 
> 
> OPTION 1
> 
> 	Use Response code 480, with new Warning header field 390 (URI
> Scheme Not Supported)
> 	for both cases when SIP or SIPS is rejected.
> 
> 	The proxy or UAS may use the warning text field to provide more
> indications, e.g.:
> 
> 	Warning: 390 isi.edu "SIPS Not Allowed"
> 	Warning: 390 isi.edu "SIPS Required"
> 
> OPTION 2
> 
> 	Use Response code 480, with 2 new Warning headera field 390
> (SIPS Not Allowed)
> 	and 391 (SIPS Required).
> 
> 	The proxy or UAS may use the warning text field to provide user,
> e.g.:
> 
> 	Warning: 390 isi.edu "Please use a SIP URI"
> 	Warning: 391 isi.edu "A secure SIPS URI is required"
> 
> 
> OPTION 1 has the characteristic of being "opaque" to Automatas, but it
> is generic
> enough to support other URI schemes. The precise reasons is described in
> human-readable
> format.
> 
> OPTON 2 has the characteristic of precisely describing the reason for
> the rejection. It
> is NOT opaque to Automatas. If new warnings are necessary for other URI
> shemes, new warning
> headers can be defined.
> 
> In both cases, existing UACs (or UACs that do not wish to have any
> special behavior based
> on this) will exhibit the desired behavior, i.e., it will look like
> "temporarily unavailable".
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 12:59:02 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Monday, July 30, 2007 10:47
> To: Audet, Francois (SC100:3055)
> Cc: IETF SIP List
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>
> I don't know what you mean about Option 1 being opaque to=20
> automata. It seems to me it provides comparable info to=20
> Option 2, giving that the UAC knows what it used in the request.

It means an Automata can not "upgrade" or "downgrade" automatically with
OPTION 1, as the indication is not a "hard" indication. It is a Warning
text
that is not standardized, and will vary by language, operators, region,
etc.

Best it can do is "record" the Warning text. That can be done
automatically
of course.



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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 13:11:34 -0700
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From: "Charles Eckel \(eckelcu\)" <eckelcu@cisco.com>
To: "Francois Audet" <audet@nortel.com>,
	"Paul Kyzivat \(pkyzivat\)" <pkyzivat@cisco.com>
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I do not have a real preference, but provided the reason for using 480 =
vs. 418 is to minimize na=EFve downgrading from SIPS to SIP, I do not =
see how option 2, resulting in:

480=20
Warning: 390 isi.edu "Please use a SIP URI"=20

helps matters.

Cheers,
Charles

> -----Original Message-----
> From: Francois Audet [mailto:audet@nortel.com]=20
> Sent: Monday, July 30, 2007 10:59 AM
> To: Paul Kyzivat (pkyzivat)
> Cc: IETF SIP List
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> =20
>=20
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> > Sent: Monday, July 30, 2007 10:47
> > To: Audet, Francois (SC100:3055)
> > Cc: IETF SIP List
> > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> > I don't know what you mean about Option 1 being opaque to=20
> > automata. It seems to me it provides comparable info to=20
> > Option 2, giving that the UAC knows what it used in the request.
>=20
> It means an Automata can not "upgrade" or "downgrade"=20
> automatically with
> OPTION 1, as the indication is not a "hard" indication. It is=20
> a Warning
> text
> that is not standardized, and will vary by language,=20
> operators, region,
> etc.
>=20
> Best it can do is "record" the Warning text. That can be done
> automatically
> of course.
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From sip-bounces@ietf.org Mon Jul 30 16:22:02 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 15:21:34 -0500
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	<49DFEF82990374449378AD1198F39F8B039FE2CD@xmb-sjc-231.amer.cisco.com>
From: "Francois Audet" <audet@nortel.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>,
	"Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>
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Existing implementations would ignore the Warning, and normal treatment =
for 480
would occur. I.e., it would be the same treatment as if the user is =
recognized as
beeing a valid user, but he's currently not logged in.

With 418, existing implementations would treat it as 400. I.e., it would =
be
as if the message itself had an error.

This may need to different UI for the user (e.g., different tone played, =
different
message displayed, etc.)

> -----Original Message-----
> From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]=20
> Sent: Monday, July 30, 2007 13:12
> To: Audet, Francois (SC100:3055); Paul Kyzivat (pkyzivat)
> Cc: IETF SIP List
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> I do not have a real preference, but provided the reason for=20
> using 480 vs. 418 is to minimize na=EFve downgrading from SIPS=20
> to SIP, I do not see how option 2, resulting in:
>=20
> 480
> Warning: 390 isi.edu "Please use a SIP URI"=20
>=20
> helps matters.
>=20
> Cheers,
> Charles
>=20
> > -----Original Message-----
> > From: Francois Audet [mailto:audet@nortel.com]
> > Sent: Monday, July 30, 2007 10:59 AM
> > To: Paul Kyzivat (pkyzivat)
> > Cc: IETF SIP List
> > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >=20
> > =20
> >=20
> > > -----Original Message-----
> > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > > Sent: Monday, July 30, 2007 10:47
> > > To: Audet, Francois (SC100:3055)
> > > Cc: IETF SIP List
> > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > > I don't know what you mean about Option 1 being opaque to=20
> automata.=20
> > > It seems to me it provides comparable info to Option 2,=20
> giving that=20
> > > the UAC knows what it used in the request.
> >=20
> > It means an Automata can not "upgrade" or "downgrade"=20
> > automatically with
> > OPTION 1, as the indication is not a "hard" indication. It is a=20
> > Warning text that is not standardized, and will vary by language,=20
> > operators, region, etc.
> >=20
> > Best it can do is "record" the Warning text. That can be done=20
> > automatically of course.
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20


_______________________________________________
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From sip-bounces@ietf.org Mon Jul 30 16:40:46 2007
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From: Eric Rescorla <ekr@networkresonance.com>
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Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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At Mon, 30 Jul 2007 09:19:04 -0700,
Michael Thomas wrote:
> 
> Eric Rescorla wrote:
> > At Fri, 27 Jul 2007 09:39:06 -0700,
> > Michael Thomas wrote:
> >   
> >> Rohan Mahy wrote:
> >>     
> >>> Michael,
> >>>
> >>> At issue here is what the default implementor is likely to do.  With a 
> >>> new 4xx, the misguided but well-meaning implementor is likely to try 
> >>> to "helpfully" "repair" the error without thinking about or 
> >>> understanding the security context.
> >>>
> >>> Using a Warning code raises the bar significantly, but still allows 
> >>> automata to at least log what happened.
> >>>       
> >> As I said, a receiver is completely at liberty to prevent the downgrade 
> >> by not
> >> accepting the downgraded request.
> >>     
> >
> > Unless, of course, someone is impersonating the receiver.
> >   
> Given how tangled up SIPS is, I really no idea what you're talking
> about, or whether it's even responsive to my suggestion. Last I heard,
> the entire raison d'etre of SIPS was that the next hop is cryptographically
> identified via TLS. I'm guessing that you're not suggesting that TLS
> is useless against impersonation attacks.

Of course not.

The point here is that if a caller automatically downgrades to SIP,
an active attacker can then intercept the request and accept it,
regardless of the receiver's preferences.

-Ekr



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To: Eric Rescorla <ekr@networkresonance.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Eric Rescorla wrote:
> At Mon, 30 Jul 2007 09:19:04 -0700,
> Michael Thomas wrote:
>   
>> Eric Rescorla wrote:
>>     
>>> At Fri, 27 Jul 2007 09:39:06 -0700,
>>> Michael Thomas wrote:
>>>   
>>>       
>>>> Rohan Mahy wrote:
>>>>     
>>>>         
>>>>> Michael,
>>>>>
>>>>> At issue here is what the default implementor is likely to do.  With a 
>>>>> new 4xx, the misguided but well-meaning implementor is likely to try 
>>>>> to "helpfully" "repair" the error without thinking about or 
>>>>> understanding the security context.
>>>>>
>>>>> Using a Warning code raises the bar significantly, but still allows 
>>>>> automata to at least log what happened.
>>>>>       
>>>>>           
>>>> As I said, a receiver is completely at liberty to prevent the downgrade 
>>>> by not
>>>> accepting the downgraded request.
>>>>     
>>>>         
>>> Unless, of course, someone is impersonating the receiver.
>>>   
>>>       
>> Given how tangled up SIPS is, I really no idea what you're talking
>> about, or whether it's even responsive to my suggestion. Last I heard,
>> the entire raison d'etre of SIPS was that the next hop is cryptographically
>> identified via TLS. I'm guessing that you're not suggesting that TLS
>> is useless against impersonation attacks.
>>     
>
> Of course not.
>
> The point here is that if a caller automatically downgrades to SIP,
> an active attacker can then intercept the request and accept it,
> regardless of the receiver's preferences.
>   

Um, so? That's the risk that the sender takes by sending stuff in the 
clear. And
where's the problem here anyway? The steps are: try SIPS, then try SIP. 
If the
first succeeds, there's no attack. So it's only in some slim window of 
when the
receiver was down, or something like that. Doesn't seem to me to be 
something
to get all exercised about, especially when you factor in all of the 
other insecurities
associated with SIPS.

       Mike


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Subject: Re: [Sip] Two minor Max-Breadth comments
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   From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>

   How does the UAC cancelling the request help.

   Surely all we end up with is the CANCEL request desperatly trying to
   catch up with all the propagated INVITE requests. I see nothing in the
   transaction model that helps it catch up, and I suspect (based on
   experience in non-SIP systems) that it never will in most people's
   implementations. It will always be pretty much the same number of steps
   behind.

By design, the total number of transactions active at any moment will
be limited to Max-Forwards times Max-Breadth.  In the canonical
example, that's 20 times 64 equals 1280.  There are 64 active
"fingers" that are depth-first-searching the tree of possible forks,
and each finger is limited to being 20 forwards long.  Of course,
there's a lot of activity at the ends of the fingers, but each
finger's actions are serial.

In the worse case, all 1280 transactions are happening on the same SIP
agent, so its work-queue can have 1280 items in it at any moment.  But
at the same time, the work-queue can have no more than 1280 items in
it.

Injecting a CANCEL at the top of the tree, the cancel may have to wait
for 1280 actions to be processed before it gets processed.  The CANCEL
stops the (unique) depth = 1 transaction from generating any more
requests, and also generates CANCELs that go to all of the depth = 2
transactions.

Each of those second-generation CANCELs may have to wait for 1280
work-items ahead of it before they are processed.  (Actually, 19 times
64 equals 1216 work-items.)  But that won't take horribly long, and
you can work out the time-bound by which time all the
second-generation CANCELs have been processed, terminated the
generation of depth = 3 requests, and caused the sending of the
third-generation CANCELs.

Etc., for depth = 3 to 20.

What saves the day is that Max-Breadth prevents any single SIP agent
from becoming totally jammed, so that any CANCEL will be processed
within a bounded amount of time.  And that Max-Forwards prevents the
fork fingers from extending beyond the limit at their tips while their
roots are being torn down by CANCELs.

Dale


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Mon, 30 Jul 2007 15:52:58 -0600
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From: "Rai, Anupam (Anupam)" <rai@avaya.com>
To: "Michael Thomas" <mat@cisco.com>,
	"Eric Rescorla" <ekr@networkresonance.com>
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An active attacker could very well DoS the SIPS attempt making it look
like "reciever was down". Hence downgrade must never bet attempted under
such condition. There must be some explict indication, like the one
discussed here, to indicate that contact does not accept SIPS.

-Anupam

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]=20
> Sent: Monday, July 30, 2007 1:55 PM
> To: Eric Rescorla
> Cc: IETF SIP List; Rohan Mahy; Francois Audet; Paul Kyzivat;=20
> Jon.Peterson@neustar.com
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> Eric Rescorla wrote:
> > At Mon, 30 Jul 2007 09:19:04 -0700,
> > Michael Thomas wrote:
> >  =20
> >> Eric Rescorla wrote:
> >>    =20
> >>> At Fri, 27 Jul 2007 09:39:06 -0700,
> >>> Michael Thomas wrote:
> >>>  =20
> >>>      =20
> >>>> Rohan Mahy wrote:
> >>>>    =20
> >>>>        =20
> >>>>> Michael,
> >>>>>
> >>>>> At issue here is what the default implementor is likely to do. =20
> >>>>> With a new 4xx, the misguided but well-meaning implementor is=20
> >>>>> likely to try to "helpfully" "repair" the error without=20
> thinking=20
> >>>>> about or understanding the security context.
> >>>>>
> >>>>> Using a Warning code raises the bar significantly, but still=20
> >>>>> allows automata to at least log what happened.
> >>>>>      =20
> >>>>>          =20
> >>>> As I said, a receiver is completely at liberty to prevent the=20
> >>>> downgrade by not accepting the downgraded request.
> >>>>    =20
> >>>>        =20
> >>> Unless, of course, someone is impersonating the receiver.
> >>>  =20
> >>>      =20
> >> Given how tangled up SIPS is, I really no idea what you're talking=20
> >> about, or whether it's even responsive to my suggestion. Last I=20
> >> heard, the entire raison d'etre of SIPS was that the next hop is=20
> >> cryptographically identified via TLS. I'm guessing that you're not=20
> >> suggesting that TLS is useless against impersonation attacks.
> >>    =20
> >
> > Of course not.
> >
> > The point here is that if a caller automatically downgrades=20
> to SIP, an=20
> > active attacker can then intercept the request and accept it,=20
> > regardless of the receiver's preferences.
> >  =20
>=20
> Um, so? That's the risk that the sender takes by sending=20
> stuff in the clear. And where's the problem here anyway? The=20
> steps are: try SIPS, then try SIP.=20
> If the
> first succeeds, there's no attack. So it's only in some slim=20
> window of when the receiver was down, or something like that.=20
> Doesn't seem to me to be something to get all exercised=20
> about, especially when you factor in all of the other=20
> insecurities associated with SIPS.
>=20
>        Mike
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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From: Michael Thomas <mat@cisco.com>
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To: "Rai, Anupam (Anupam)" <rai@avaya.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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And the way to deal with this is some obscure ietf nanny text in some 
document
that may or may not be read by developers? I'm sorry, but I don't find 
this a very
compelling way to go about fixing this kind of problem.

But if you're going to do this at all, at least instruct them that they 
should only
try the SIP downgrade at first contact, so that you don't have to keep 
exposing
yourself on subsequent invitations. Telling them to not do it at all is 
about the
worst advise because it will be ignored.

       Mike



Rai, Anupam (Anupam) wrote:
> An active attacker could very well DoS the SIPS attempt making it look
> like "reciever was down". Hence downgrade must never bet attempted under
> such condition. There must be some explict indication, like the one
> discussed here, to indicate that contact does not accept SIPS.
>
> -Anupam
>
>   
>> -----Original Message-----
>> From: Michael Thomas [mailto:mat@cisco.com] 
>> Sent: Monday, July 30, 2007 1:55 PM
>> To: Eric Rescorla
>> Cc: IETF SIP List; Rohan Mahy; Francois Audet; Paul Kyzivat; 
>> Jon.Peterson@neustar.com
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> Eric Rescorla wrote:
>>     
>>> At Mon, 30 Jul 2007 09:19:04 -0700,
>>> Michael Thomas wrote:
>>>   
>>>       
>>>> Eric Rescorla wrote:
>>>>     
>>>>         
>>>>> At Fri, 27 Jul 2007 09:39:06 -0700,
>>>>> Michael Thomas wrote:
>>>>>   
>>>>>       
>>>>>           
>>>>>> Rohan Mahy wrote:
>>>>>>     
>>>>>>         
>>>>>>             
>>>>>>> Michael,
>>>>>>>
>>>>>>> At issue here is what the default implementor is likely to do.  
>>>>>>> With a new 4xx, the misguided but well-meaning implementor is 
>>>>>>> likely to try to "helpfully" "repair" the error without 
>>>>>>>               
>> thinking 
>>     
>>>>>>> about or understanding the security context.
>>>>>>>
>>>>>>> Using a Warning code raises the bar significantly, but still 
>>>>>>> allows automata to at least log what happened.
>>>>>>>       
>>>>>>>           
>>>>>>>               
>>>>>> As I said, a receiver is completely at liberty to prevent the 
>>>>>> downgrade by not accepting the downgraded request.
>>>>>>     
>>>>>>         
>>>>>>             
>>>>> Unless, of course, someone is impersonating the receiver.
>>>>>   
>>>>>       
>>>>>           
>>>> Given how tangled up SIPS is, I really no idea what you're talking 
>>>> about, or whether it's even responsive to my suggestion. Last I 
>>>> heard, the entire raison d'etre of SIPS was that the next hop is 
>>>> cryptographically identified via TLS. I'm guessing that you're not 
>>>> suggesting that TLS is useless against impersonation attacks.
>>>>     
>>>>         
>>> Of course not.
>>>
>>> The point here is that if a caller automatically downgrades 
>>>       
>> to SIP, an 
>>     
>>> active attacker can then intercept the request and accept it, 
>>> regardless of the receiver's preferences.
>>>   
>>>       
>> Um, so? That's the risk that the sender takes by sending 
>> stuff in the clear. And where's the problem here anyway? The 
>> steps are: try SIPS, then try SIP. 
>> If the
>> first succeeds, there's no attack. So it's only in some slim 
>> window of when the receiver was down, or something like that. 
>> Doesn't seem to me to be something to get all exercised 
>> about, especially when you factor in all of the other 
>> insecurities associated with SIPS.
>>
>>        Mike
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip 
>> Use sipping@ietf.org for new developments on the application of sip
>>
>>     


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From sip-bounces@ietf.org Mon Jul 30 18:10:52 2007
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Date: Mon, 30 Jul 2007 15:08:34 -0700
From: Eric Rescorla <ekr@networkresonance.com>
To: Michael Thomas <mat@cisco.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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At Mon, 30 Jul 2007 13:54:41 -0700,
Michael Thomas wrote:
> 
> Eric Rescorla wrote:
> > At Mon, 30 Jul 2007 09:19:04 -0700,
> > Michael Thomas wrote:
> >   
> >> Eric Rescorla wrote:
> >>     
> >>> At Fri, 27 Jul 2007 09:39:06 -0700,
> >>> Michael Thomas wrote:
> >>>   
> >>>       
> >>>> Rohan Mahy wrote:
> >>>>     
> >>>>         
> >>>>> Michael,
> >>>>>
> >>>>> At issue here is what the default implementor is likely to do.  With a 
> >>>>> new 4xx, the misguided but well-meaning implementor is likely to try 
> >>>>> to "helpfully" "repair" the error without thinking about or 
> >>>>> understanding the security context.
> >>>>>
> >>>>> Using a Warning code raises the bar significantly, but still allows 
> >>>>> automata to at least log what happened.
> >>>>>       
> >>>>>           
> >>>> As I said, a receiver is completely at liberty to prevent the downgrade 
> >>>> by not
> >>>> accepting the downgraded request.
> >>>>     
> >>>>         
> >>> Unless, of course, someone is impersonating the receiver.
> >>>   
> >>>       
> >> Given how tangled up SIPS is, I really no idea what you're talking
> >> about, or whether it's even responsive to my suggestion. Last I heard,
> >> the entire raison d'etre of SIPS was that the next hop is cryptographically
> >> identified via TLS. I'm guessing that you're not suggesting that TLS
> >> is useless against impersonation attacks.
> >>     
> >
> > Of course not.
> >
> > The point here is that if a caller automatically downgrades to SIP,
> > an active attacker can then intercept the request and accept it,
> > regardless of the receiver's preferences.
> >   
> 
> Um, so? That's the risk that the sender takes by sending stuff in the 
> clear. And
> where's the problem here anyway? The steps are: try SIPS, then try SIP. 
> If the
> first succeeds, there's no attack. So it's only in some slim window of 
> when the
> receiver was down, or something like that. Doesn't seem to me to be 
> something
> to get all exercised about, especially when you factor in all of the 
> other insecurities
> associated with SIPS.

I'm not saying it's something huge to get exercised about. However, since
the question is what behavior to define here, we might as well get it
right.

-Ekr


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Thread-Topic: Warn-Codes and draft-ietf-sip-sips
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From: "Francois Audet" <audet@nortel.com>
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Now that we have everybody exited about the prospect of using a
Warn-Code
for "SIPS Not Allowed" and "SIP Required" with Response 480, instead of
using
new response codes, here is a quote from 27.2/RFC 3261.

   Warning codes provide information supplemental to the status code in
   SIP response messages when the failure of the transaction results
   from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.

My reading of this is that Warn-Codes are ONLY usable for SDP errors.

Doesn't this disqualify the idea of using a Warn-Code for SIP/SIPS URI
problems??????

If so, aren't we back to 418/419, or 418+New header (Allow/Require), or=20
480+Response text?


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From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: "Francois Audet" <audet@nortel.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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With a new 4xx response, existing implementations would treat it as 400
anyway.

In any case, option 1 is good enough, but I am not a fan of using reason
phrases as indicators to humans. I call someone is Japan and get a Japanese
string telling me 'SIPS required'. I would have no clue what its saying.

Hisham


On 31/07/07, Francois Audet <audet@nortel.com> wrote:
>
> Existing implementations would ignore the Warning, and normal treatment
> for 480
> would occur. I.e., it would be the same treatment as if the user is
> recognized as
> beeing a valid user, but he's currently not logged in.
>
> With 418, existing implementations would treat it as 400. I.e., it would
> be
> as if the message itself had an error.
>
> This may need to different UI for the user (e.g., different tone played,
> different
> message displayed, etc.)
>
> > -----Original Message-----
> > From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
> > Sent: Monday, July 30, 2007 13:12
> > To: Audet, Francois (SC100:3055); Paul Kyzivat (pkyzivat)
> > Cc: IETF SIP List
> > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> > I do not have a real preference, but provided the reason for
> > using 480 vs. 418 is to minimize na=EFve downgrading from SIPS
> > to SIP, I do not see how option 2, resulting in:
> >
> > 480
> > Warning: 390 isi.edu "Please use a SIP URI"
> >
> > helps matters.
> >
> > Cheers,
> > Charles
> >
> > > -----Original Message-----
> > > From: Francois Audet [mailto:audet@nortel.com]
> > > Sent: Monday, July 30, 2007 10:59 AM
> > > To: Paul Kyzivat (pkyzivat)
> > > Cc: IETF SIP List
> > > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > > > Sent: Monday, July 30, 2007 10:47
> > > > To: Audet, Francois (SC100:3055)
> > > > Cc: IETF SIP List
> > > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > > >
> > > > I don't know what you mean about Option 1 being opaque to
> > automata.
> > > > It seems to me it provides comparable info to Option 2,
> > giving that
> > > > the UAC knows what it used in the request.
> > >
> > > It means an Automata can not "upgrade" or "downgrade"
> > > automatically with
> > > OPTION 1, as the indication is not a "hard" indication. It is a
> > > Warning text that is not standardized, and will vary by language,
> > > operators, region, etc.
> > >
> > > Best it can do is "record" the Warning text. That can be done
> > > automatically of course.
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> > >
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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<div>With a new 4xx response, existing implementations would treat it as 40=
0 anyway.</div>
<div>&nbsp;</div>
<div>In any case, option 1 is good enough, but I am not a fan of using reas=
on phrases as indicators to humans. I call someone is Japan and get a Japan=
ese string telling me &#39;SIPS required&#39;. I would have no clue what it=
s saying.
</div>
<div>&nbsp;</div>
<div>Hisham<br><br>&nbsp;</div>
<div><span class=3D"gmail_quote">On 31/07/07, <b class=3D"gmail_sendername"=
>Francois Audet</b> &lt;<a href=3D"mailto:audet@nortel.com">audet@nortel.co=
m</a>&gt; wrote:</span>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Existing implementations would i=
gnore the Warning, and normal treatment for 480<br>would occur. I.e., it wo=
uld be the same treatment as if the user is recognized as
<br>beeing a valid user, but he&#39;s currently not logged in.<br><br>With =
418, existing implementations would treat it as 400. I.e., it would be<br>a=
s if the message itself had an error.<br><br>This may need to different UI =
for the user (
e.g., different tone played, different<br>message displayed, etc.)<br><br>&=
gt; -----Original Message-----<br>&gt; From: Charles Eckel (eckelcu) [mailt=
o:<a href=3D"mailto:eckelcu@cisco.com">eckelcu@cisco.com</a>]<br>&gt; Sent:=
 Monday, July 30, 2007 13:12
<br>&gt; To: Audet, Francois (SC100:3055); Paul Kyzivat (pkyzivat)<br>&gt; =
Cc: IETF SIP List<br>&gt; Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs=
. 418<br>&gt;<br>&gt; I do not have a real preference, but provided the rea=
son for
<br>&gt; using 480 vs. 418 is to minimize na=EFve downgrading from SIPS<br>=
&gt; to SIP, I do not see how option 2, resulting in:<br>&gt;<br>&gt; 480<b=
r>&gt; Warning: 390 <a href=3D"http://isi.edu">isi.edu</a> &quot;Please use=
 a SIP URI&quot;
<br>&gt;<br>&gt; helps matters.<br>&gt;<br>&gt; Cheers,<br>&gt; Charles<br>=
&gt;<br>&gt; &gt; -----Original Message-----<br>&gt; &gt; From: Francois Au=
det [mailto:<a href=3D"mailto:audet@nortel.com">audet@nortel.com</a>]<br>
&gt; &gt; Sent: Monday, July 30, 2007 10:59 AM<br>&gt; &gt; To: Paul Kyziva=
t (pkyzivat)<br>&gt; &gt; Cc: IETF SIP List<br>&gt; &gt; Subject: RE: [Sip]=
 draft-ietf-sip-sips-05: 480 vs. 418<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt;
<br>&gt; &gt; &gt; -----Original Message-----<br>&gt; &gt; &gt; From: Paul =
Kyzivat [mailto:<a href=3D"mailto:pkyzivat@cisco.com">pkyzivat@cisco.com</a=
>]<br>&gt; &gt; &gt; Sent: Monday, July 30, 2007 10:47<br>&gt; &gt; &gt; To=
: Audet, Francois (SC100:3055)
<br>&gt; &gt; &gt; Cc: IETF SIP List<br>&gt; &gt; &gt; Subject: Re: [Sip] d=
raft-ietf-sip-sips-05: 480 vs. 418<br>&gt; &gt; &gt;<br>&gt; &gt; &gt; I do=
n&#39;t know what you mean about Option 1 being opaque to<br>&gt; automata.
<br>&gt; &gt; &gt; It seems to me it provides comparable info to Option 2,<=
br>&gt; giving that<br>&gt; &gt; &gt; the UAC knows what it used in the req=
uest.<br>&gt; &gt;<br>&gt; &gt; It means an Automata can not &quot;upgrade&=
quot; or &quot;downgrade&quot;
<br>&gt; &gt; automatically with<br>&gt; &gt; OPTION 1, as the indication i=
s not a &quot;hard&quot; indication. It is a<br>&gt; &gt; Warning text that=
 is not standardized, and will vary by language,<br>&gt; &gt; operators, re=
gion, etc.
<br>&gt; &gt;<br>&gt; &gt; Best it can do is &quot;record&quot; the Warning=
 text. That can be done<br>&gt; &gt; automatically of course.<br>&gt; &gt;<=
br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt; ____________________________________=
___________
<br>&gt; &gt; Sip mailing list&nbsp;&nbsp;<a href=3D"https://www1.ietf.org/=
mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt=
; &gt; This list is for NEW development of the core SIP Protocol Use<br>&gt=
; &gt; <a href=3D"mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip Use<br>&g=
t; &gt; <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new de=
velopments on the application of sip<br>&gt; &gt;<br>&gt;<br><br><br>______=
_________________________________________
<br>Sip mailing list&nbsp;&nbsp;<a href=3D"https://www1.ietf.org/mailman/li=
stinfo/sip">https://www1.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-i=
mplementors@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 developments=
 on the application of sip<br></blockquote></div><br>

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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1300011482==--





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Date: Tue, 31 Jul 2007 10:54:46 +1000
From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: "Francois Audet" <audet@nortel.com>
Subject: Re: [Sip] Warn-Codes and draft-ietf-sip-sips
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Like I said in an earlier email, Reason phrases are not good enough for me,
unless my phone has a translation for every language on Earth.

It is much preferred that my phone looks at the error code and presents me
with a localised reason phrase.

Hisham


On 31/07/07, Francois Audet <audet@nortel.com> wrote:
>
> Now that we have everybody exited about the prospect of using a
> Warn-Code
> for "SIPS Not Allowed" and "SIP Required" with Response 480, instead of
> using
> new response codes, here is a quote from 27.2/RFC 3261.
>
>   Warning codes provide information supplemental to the status code in
>   SIP response messages when the failure of the transaction results
>   from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.
>
> My reading of this is that Warn-Codes are ONLY usable for SDP errors.
>
> Doesn't this disqualify the idea of using a Warn-Code for SIP/SIPS URI
> problems??????
>
> If so, aren't we back to 418/419, or 418+New header (Allow/Require), or
> 480+Response text?
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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<div>Like I said in an earlier email, Reason phrases are not good enough for me, unless my phone has a translation for every language on Earth.</div>
<div>&nbsp;</div>
<div>It is much preferred that my phone looks at the error code and presents me with a localised reason phrase.</div>
<div>&nbsp;</div>
<div>Hisham<br><br>&nbsp;</div>
<div><span class="gmail_quote">On 31/07/07, <b class="gmail_sendername">Francois Audet</b> &lt;<a href="mailto:audet@nortel.com">audet@nortel.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Now that we have everybody exited about the prospect of using a<br>Warn-Code<br>for &quot;SIPS Not Allowed&quot; and &quot;SIP Required&quot; with Response 480, instead of
<br>using<br>new response codes, here is a quote from 27.2/RFC 3261.<br><br>&nbsp;&nbsp;Warning codes provide information supplemental to the status code in<br>&nbsp;&nbsp;SIP response messages when the failure of the transaction results<br>
&nbsp;&nbsp;from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.<br><br>My reading of this is that Warn-Codes are ONLY usable for SDP errors.<br><br>Doesn&#39;t this disqualify the idea of using a Warn-Code for SIP/SIPS URI
<br>problems??????<br><br>If so, aren&#39;t we back to 418/419, or 418+New header (Allow/Require), or<br>480+Response text?<br><br><br>_______________________________________________<br>Sip mailing list&nbsp;&nbsp;<a href="https://www1.ietf.org/mailman/listinfo/sip">
https://www1.ietf.org/mailman/listinfo/sip</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip
<br>Use <a href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br></blockquote></div><br>

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From sip-bounces@ietf.org Mon Jul 30 21:37:22 2007
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Date: Tue, 31 Jul 2007 09:36:24 +0800
From: Qian Sun <sunqian@huawei.com>
Subject: RE: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides
	forbody (MIME) handling)
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Hi, Dean 

Sorry for late response. Do you mean "advertisement" content always is needed to 'render'? So a single content disposition "advertisement" is enough.

In addition, some content body (e.g. Geolocation or advertisement) may be inserted or modified by intermediary. Could we specify a common way for indicating the source of inserted content body? For example, define a new MIME header: Content-Insertedby. A B2BUA may insert a content body including AD like this:

Content-Type: text/plain
Content-Disposition: advertisement
Content-Insertedby: AD-Agent <sip: adagent@example.com>

McDonald, i'm lovin' it!

Opinion?

Cheers,
Qian

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com] 
Sent: Friday, July 20, 2007 6:45 PM
To: Qian Sun
Cc: 'Paul Kyzivat'; 'Gonzalo Camarillo'; 'sip'
Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides forbody (MIME) handling)

Qian Sun wrote:
> If we define a new Content Disposition type "advertisement" so that
> the operator or service provider may insert advertisement in the all
> kinds of message body. And we want the advertisement content 'render'
> to the users. Could we use multiple content dispositions
> ("advertisement" and 'render') for a single content part? Or only
> define a new Content Disposition Parameters "advertisement" instead?

So is there a need to ever use "advertisement" in conjunction with 
something besides "render"?

--
Dean





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I'd rather not institutionalize advertising in ietf standards.
BUT, if we can get an agreement that all advertisements, of any sort, 
MUST be labeled with the "advertisement" content-disposition.

	Paul

Qian Sun wrote:
> Hi, Dean 
> 
> Sorry for late response. Do you mean "advertisement" content always is needed to 'render'? So a single content disposition "advertisement" is enough.
> 
> In addition, some content body (e.g. Geolocation or advertisement) may be inserted or modified by intermediary. Could we specify a common way for indicating the source of inserted content body? For example, define a new MIME header: Content-Insertedby. A B2BUA may insert a content body including AD like this:
> 
> Content-Type: text/plain
> Content-Disposition: advertisement
> Content-Insertedby: AD-Agent <sip: adagent@example.com>
> 
> McDonald, i'm lovin' it!
> 
> Opinion?
> 
> Cheers,
> Qian
> 
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: Friday, July 20, 2007 6:45 PM
> To: Qian Sun
> Cc: 'Paul Kyzivat'; 'Gonzalo Camarillo'; 'sip'
> Subject: Re: Content-Disposition in multiparts WAS (Re: [Sip] Draft slides forbody (MIME) handling)
> 
> Qian Sun wrote:
>> If we define a new Content Disposition type "advertisement" so that
>> the operator or service provider may insert advertisement in the all
>> kinds of message body. And we want the advertisement content 'render'
>> to the users. Could we use multiple content dispositions
>> ("advertisement" and 'render') for a single content part? Or only
>> define a new Content Disposition Parameters "advertisement" instead?
> 
> So is there a need to ever use "advertisement" in conjunction with 
> something besides "render"?
> 
> --
> Dean
> 
> 


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At 10:54 AM 7/30/2007, Francois Audet wrote:
>RFC 3261, p. 181:
>
>         "The first digit  of warning codes beginning with "3" indicates
>warnings
>         specific to SIP."
>
>Then it explains within the 3XX space, how they are used.

btw -- for location-conveyance-07, I created 15 new Warning codes (#s 
701 through 712, then 720, 721, and 722).  All this at the suggestion 
of Henning (who knows something about 3261 text...  ;-)

btw II - I have since replaced the Warning header indication with a 
new Geolocation-Error header, but this was so I could stuff more into 
the new header than the Warning header would allow.

In all this, I did not get one person claiming Warning codes were 
limited to 3XX.  Henning merely said the "first batch of codes were 
in the 300s, and that there was nothing limiting a new ID from 
defining new codes in any other number range"...


> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Saturday, July 28, 2007 18:25
> > To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat
> > Paul; Mahy Rohan
> > Cc: IETF SIP List; Jon.Peterson@neustar.com
> > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> > A 399 Miscellaneous?  That doesn't make sense...
> >
> > Warning codes are not limited to 3XX, so if a new Warning
> > code is what Rohan wants to convey more information in the
> > response (which I believe should be done one way or another)
> > - it can be almost any value other than 300-379 or 399 IMO.
> >
> > At 10:32 AM 7/27/2007, Francois Audet wrote:
> > >On the Warning header...
> > >
> > >Values 300-379 are reserved for SDP-related stuff.
> > >
> > >Values 380-389 are unassiged (but there is no text on it).
> > >
> > >Values 390-398 are unassigned (but don't fall into the SDP related
> > >stuff.
> > >
> > >399 means Miscellaneous and the text can be provided to the user.
> > >It further says that Automata MUST NOT take action based on this.
> > >
> > >Rohan, are you advocating using 399, or defining a new code from the
> > >390-398 space?
> > >
> > > > -----Original Message-----
> > > > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com]
> > On Behalf Of
> > > > Alan Hawrylyshen
> > > > Sent: Friday, July 27, 2007 08:27
> > > > To: Kyzivat Paul; Mahy Rohan
> > > > Cc: Audet, Francois (SC100:3055); IETF SIP List;
> > > > Jon.Peterson@neustar.com
> > > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > > >
> > > >
> > > > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> > > >
> > > > > I understand that if the user entered a sips URI then it
> > > > should be the
> > > > > user that must decide to downgrade. But if the user didn't know
> > > > > whether to use sip or sips in the first place, and the UA
> > > > decides to
> > > > > try sips first then I see no problem in the UA having a policy
> > > > > that causes it to downgrade.
> > > >
> > > > I was under the impression (based on meeting discussion) that :
> > > > 1 - the downgrade was undesirable because it reveals
> > > > (possibly) information about the targeted party in the clear, and;
> > > > 2 - The 480 with a Warning header was an option to provide
> > > > automata- friendly indication of the failure reason.
> > > >
> > > > Alan Hawrylyshen
> > > >
> > > > a l a n a t p o l y p h a s e d o t c a
> > > >
> > > >
> > > >
> > >
> > >
> > >_______________________________________________
> > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >This list is for NEW development of the core SIP Protocol Use
> > >sip-implementors@cs.columbia.edu for questions on current sip Use
> > >sipping@ietf.org for new developments on the application of sip
> >


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From: "Robins, Ellen" <Ellen.Robins@team.telstra.com>
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 Dear all,
Please help to confirm which bandwidth parameter shall be used by Video
Terminal Client for a SIP Client to SIP Client call case, i.e. between 2
parties not a three-way conference.

   Bandwidth

   b=3D<modifier>:<bandwidth-value>

   o This specifies the proposed bandwidth to be used by the session or
     media, and is optional.

   o <bandwidth-value> is in kilobits per second

   o <modifier> is a single alphanumeric word giving the meaning of the
     bandwidth figure.

   o Two modifiers are initially defined:

   CT Conference Total: An implicit maximum bandwidth is associated with
     each TTL on the Mbone or within a particular multicast
     administrative scope region (the Mbone bandwidth vs. TTL limits are
     given in the MBone FAQ). If the bandwidth of a session or media in
     a session is different from the bandwidth implicit from the scope,
     a `b=3DCT:...' line should be supplied for the session giving the
     proposed upper limit to the bandwidth used. The primary purpose of
     this is to give an approximate idea as to whether two or more
     conferences can co-exist simultaneously.

   AS Application-Specific Maximum: The bandwidth is interpreted to be
     application-specific, i.e., will be the application's concept of
     maximum bandwidth.  Normally this will coincide with what is set on
     the application's "maximum bandwidth" control if applicable.

     Note that CT gives a total bandwidth figure for all the media at
     all sites.  AS gives a bandwidth figure for a single media at a
     single site, although there may be many sites sending
     simultaneously.


Thanks and kind regards / Ellen Robins
ViG Technology Manager
Services Enablers
Wireless Engineering & Operations
Telstra Operations
Telstra Corporation Limited
 =20
Ph 03 8627 8981
Mob 0428 309 250
Fax 03 9620 7864
Mail ellen.robins@team.telstra.com=20

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From: Aki Niemi <aki.niemi@nokia.com>
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Subject: [Sip] subnot-etags: suppressing NOTIFY interactions with SBCs and
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Hi all,

This is a potential show-stopper issue that Rohan brought up in the SIP
WG session last week.

Currently, subnot-etags offers two modes of operation. In the first, the
body of the SUBSCRIBE-generated NOTIFY is suppressed, and in the other,
the entire NOTIFY is suppressed, and a new response 204 (No
Notification) is generated instead.

The latter is a considerable change to how SIP events works, and might
cause a middlebox that tracks the SIP dialog to basically barf, thinking
the dialog expires when it in fact gets refreshed.

Do folks think this is a problem? It would be great if some SBC vendors
could check what their implementation does with SUBSCRIBE dialogs.

Personally, I'd hate to hack around boxes that people claim are doing
something fishy. But if this is a real problem, then subnot-etags would
basically need to be reduced to just suppressing bodies of NOTIFYs.

Cheers,
Aki


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Tue Jul 31 03:53:35 2007
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Subject: [Sip] Some comments for location-conveyance-08
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Hi,

I had a review of draft-ietf-sip-location-conveyance-08 and
have the following comments to it. I have not followed up
the email discussions for this draft so please apologize
if any of my comments have already been raised.

1. Possible values within the inserted-by parameter

What kind of value should the UAC enter into the "inserted-by"
parameter if it does not have a hostname ? Is e.g. the local IP=20
address of the UAC OK ?

The BNF syntax refers to "host-id" when describing the contents
of the "inserted-by" parameter. However I did not find anywhere
any formal BNF definition to the "host-id" (either in this draft
or RFC 3261), thus it is not clear if it only refers to a hostname
or could it also refer to e.g. an IP address or some other=20
cryptographically constructed random ID allocated by the UAC.

2. Handling of 424 responses

In my opinion the handling the 424 responses is not well specified:

UAC:=20

   The Geolocation-Error has a header parameter indicating which entity
   inserted the location pertaining to this error.  This is in the form
   of a host-id.  A UAC receiving this 424 should review this=20
   "inserter=3D" locationErrorValue parameter to see if it indicates =
this
   UAC.  If it does not, the 424 should be ignored.=20

First of all the Geolocation-Error header might contain multiple=20
locationErrorValue entries. It is not clear in the normative text
for UAC that the UAC is expected to search through all of those to
check if it can find any entries targetted to itself. Further on
if there is no entries for the UAC, shall the UAC really ignore
the 424 and continue retransmitting the request or handle it as
4xx response and stop retransmissions ?

What if it was the proxy to insert Geolocation header into the
request and the UAS would return 424 response ? There is no=20
normative text for the proxy to take this response into account
even if the Geolocation-Error header would refer to the proxy.

3. Usage of geolocation tag in the Supported header

Usage of the geolocation tag in the Supported header is not
specified according to the guidelines of RFC 3261. There were=20
similar issues raised for SIP Outbound recently and the agreement
was to fix them:
http://www1.ietf.org/mail-archive/web/sip/current/msg20045.html

RFC 3261:

   If the UAC supports extensions to SIP that can be applied by the
   server to the response, the UAC SHOULD include a Supported header
   field in the request listing the option tags (Section 19.2) for those
   extensions.

draft-ietf-sip-location-conveyance-08:

   Whenever a UA wants to=20
   indicate support for this SIP extension, the geolocation option tag=20
   is included in a Supported header of the SIP message. =20

   The UAC supporting this=20
   extension MUST include a Supported header with the geolocation=20
   option tag.

   The geolocation option-tag is inserted in a Supported header by a=20
   UAC to provide an indication of support for this extension.  The=20
   presence of the geolocation option tag in a Supported header without
   a Geolocation header field in the same message informs a receiving=20
   SIP element the UAC understands this extension, but it does not know
   or wish to convey its location at this time.  Certain scenarios=20
   exist (location-based retargeting) in which location is required in
   a SIP request in order to retarget the message properly.  This=20
   affects how a UAS or SIP server processes to such a request.

The semantics of the geolocation tag within the Supported header seems
to be vaguely defined. The presence of this tag does not indicate the
UAS (or proxy) any extensions that the server could apply to its
response.
According to RFC 3261 that would however be the reason why to include an
option tag into the Supported header of a request. All the use cases
given in the draft for the geolocation tag justify its usage within
the Require header but there seems to be no clear reason why the UAC
would add it into the Supported header of a request. Support for the
extension can also be inferred from the presence of Geolocation header.
If it is not present but the a proxy would require it for retargeting,=20
the correct behaviour would be the proxy to send 421 Extension Required
response.

   A UAS MAY include a geolocation option-tag in the Supported header=20
   of a response, indicating if does understand this extension.

Similarily for the UAS no clear reasons exist why to include the=20
Supported header into its response. =20

Nit:

4.1 Location-by-value

   Geolocation: <cid:target123@atlanta.example.com>=20

   --boundary1

   Content-Type: application/pidf+xml
   Content-ID: alice123@atlanta.example.com

In Geolocation header the Content ID begins with "target123"
even if within the MIME entity the ID begins with "alice123".


Regards,

Erkki Koivusalo


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I don't think this is a real problem. It is a *change*, no worse than 
other changes we make from time to time. It won't bother the sorts of 
middleboxes (proxies) that we have clear standards for. If it bothers 
B2BUAs then they will need to learn how to deal with it.

	Paul

Aki Niemi wrote:
> Hi all,
> 
> This is a potential show-stopper issue that Rohan brought up in the SIP
> WG session last week.
> 
> Currently, subnot-etags offers two modes of operation. In the first, the
> body of the SUBSCRIBE-generated NOTIFY is suppressed, and in the other,
> the entire NOTIFY is suppressed, and a new response 204 (No
> Notification) is generated instead.
> 
> The latter is a considerable change to how SIP events works, and might
> cause a middlebox that tracks the SIP dialog to basically barf, thinking
> the dialog expires when it in fact gets refreshed.
> 
> Do folks think this is a problem? It would be great if some SBC vendors
> could check what their implementation does with SUBSCRIBE dialogs.
> 
> Personally, I'd hate to hack around boxes that people claim are doing
> something fishy. But if this is a real problem, then subnot-etags would
> basically need to be reduced to just suppressing bodies of NOTIFYs.
> 
> Cheers,
> Aki
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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I'm worried that, if you eliminate suppression of NOTIFY itself, the 
primary value of this draft (reduction of message traffic) is lost. I'd 
like to keep this capability if we can.

-Jonathan R.

Aki Niemi wrote:
> Hi all,
> 
> This is a potential show-stopper issue that Rohan brought up in the SIP
> WG session last week.
> 
> Currently, subnot-etags offers two modes of operation. In the first, the
> body of the SUBSCRIBE-generated NOTIFY is suppressed, and in the other,
> the entire NOTIFY is suppressed, and a new response 204 (No
> Notification) is generated instead.
> 
> The latter is a considerable change to how SIP events works, and might
> cause a middlebox that tracks the SIP dialog to basically barf, thinking
> the dialog expires when it in fact gets refreshed.
> 
> Do folks think this is a problem? It would be great if some SBC vendors
> could check what their implementation does with SUBSCRIBE dialogs.
> 
> Personally, I'd hate to hack around boxes that people claim are doing
> something fishy. But if this is a real problem, then subnot-etags would
> basically need to be reduced to just suppressing bodies of NOTIFYs.
> 
> Cheers,
> Aki
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Tue, 31 Jul 2007 08:49:24 -0500
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From: "Francois Audet" <audet@nortel.com>
To: "James M. Polk" <jmpolk@cisco.com>, "Alan Hawrylyshen" <alan@polyphase.ca>,
	"Kyzivat Paul" <pkyzivat@cisco.com>, "Mahy Rohan" <rohan@ekabal.com>
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Ok, that's fine with me. It's just a glitch in 3261 then. I'd rather we
stick with the Warn-Codes approach.

Did you assign a purpose for 7XX-series warn-codes yet? I'm wondering
what
range should be used for this purpose.

James, I didn't see your codes on IANA's web site. What values are you
using?
I don't want to overlap.


> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]=20
> Sent: Monday, July 30, 2007 21:44
> To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat=20
> Paul; Mahy Rohan
> Cc: IETF SIP List; Jon.Peterson@neustar.com
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> At 10:54 AM 7/30/2007, Francois Audet wrote:
> >RFC 3261, p. 181:
> >
> >         "The first digit  of warning codes beginning with "3"=20
> >indicates warnings
> >         specific to SIP."
> >
> >Then it explains within the 3XX space, how they are used.
>=20
> btw -- for location-conveyance-07, I created 15 new Warning codes (#s
> 701 through 712, then 720, 721, and 722).  All this at the=20
> suggestion of Henning (who knows something about 3261 text...  ;-)
>=20
> btw II - I have since replaced the Warning header indication=20
> with a new Geolocation-Error header, but this was so I could=20
> stuff more into the new header than the Warning header would allow.
>=20
> In all this, I did not get one person claiming Warning codes=20
> were limited to 3XX.  Henning merely said the "first batch of=20
> codes were in the 300s, and that there was nothing limiting a=20
> new ID from defining new codes in any other number range"...
>=20
>=20
> > > -----Original Message-----
> > > From: James M. Polk [mailto:jmpolk@cisco.com]
> > > Sent: Saturday, July 28, 2007 18:25
> > > To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat Paul;=20
> > > Mahy Rohan
> > > Cc: IETF SIP List; Jon.Peterson@neustar.com
> > > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > >
> > > A 399 Miscellaneous?  That doesn't make sense...
> > >
> > > Warning codes are not limited to 3XX, so if a new Warning code is=20
> > > what Rohan wants to convey more information in the=20
> response (which I=20
> > > believe should be done one way or another)
> > > - it can be almost any value other than 300-379 or 399 IMO.
> > >
> > > At 10:32 AM 7/27/2007, Francois Audet wrote:
> > > >On the Warning header...
> > > >
> > > >Values 300-379 are reserved for SDP-related stuff.
> > > >
> > > >Values 380-389 are unassiged (but there is no text on it).
> > > >
> > > >Values 390-398 are unassigned (but don't fall into the=20
> SDP related=20
> > > >stuff.
> > > >
> > > >399 means Miscellaneous and the text can be provided to the user.
> > > >It further says that Automata MUST NOT take action based on this.
> > > >
> > > >Rohan, are you advocating using 399, or defining a new code from=20
> > > >the
> > > >390-398 space?
> > > >
> > > > > -----Original Message-----
> > > > > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com]
> > > On Behalf Of
> > > > > Alan Hawrylyshen
> > > > > Sent: Friday, July 27, 2007 08:27
> > > > > To: Kyzivat Paul; Mahy Rohan
> > > > > Cc: Audet, Francois (SC100:3055); IETF SIP List;=20
> > > > > Jon.Peterson@neustar.com
> > > > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > > > >
> > > > >
> > > > > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> > > > >
> > > > > > I understand that if the user entered a sips URI then it
> > > > > should be the
> > > > > > user that must decide to downgrade. But if the user didn't=20
> > > > > > know whether to use sip or sips in the first place,=20
> and the UA
> > > > > decides to
> > > > > > try sips first then I see no problem in the UA=20
> having a policy=20
> > > > > > that causes it to downgrade.
> > > > >
> > > > > I was under the impression (based on meeting=20
> discussion) that :
> > > > > 1 - the downgrade was undesirable because it reveals
> > > > > (possibly) information about the targeted party in the clear,=20
> > > > > and;
> > > > > 2 - The 480 with a Warning header was an option to provide
> > > > > automata- friendly indication of the failure reason.
> > > > >
> > > > > Alan Hawrylyshen
> > > > >
> > > > > a l a n a t p o l y p h a s e d o t c a
> > > > >
> > > > >
> > > > >
> > > >
> > > >
> > > >_______________________________________________
> > > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >This list is for NEW development of the core SIP Protocol Use=20
> > > >sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > >sipping@ietf.org for new developments on the application of sip
> > >
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
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Fran=E7ois,

I think the IANA paragraph you described is too restrictive.  The =20
text in RFC3261 section 20.43 seems perfectly fine with registering =20
non-SDP errors.  I would allocate warn-code 380 "No SIPS contacts =20
registered".

thanks,
-rohan

On Jul 30, 2007, at 3:23 PM, Francois Audet wrote:

> Now that we have everybody exited about the prospect of using a
> Warn-Code
> for "SIPS Not Allowed" and "SIP Required" with Response 480, =20
> instead of
> using
> new response codes, here is a quote from 27.2/RFC 3261.
>
>    Warning codes provide information supplemental to the status =20
> code in
>    SIP response messages when the failure of the transaction results
>    from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.
>
> My reading of this is that Warn-Codes are ONLY usable for SDP errors.
>
> Doesn't this disqualify the idea of using a Warn-Code for SIP/SIPS URI
> problems??????
>
> If so, aren't we back to 418/419, or 418+New header (Allow/=20
> Require), or
> 480+Response text?



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Subject: [Sip] Draft Liaision response to SG13 on Emergency
	Telecommunications
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Hi

This is the draft for a liaison response to be sent to SG 13 of the=20
ITU-T from the TSVWG, NSIS, SIP and SIPPING WGs. Please provide any=20
comments at the latest by the 14th of August. After that we will address=20
any raised comments and send the liaison to ITU-T before the end of Augus=
t.

Submission
Date:		<tbd>, 2007

From: 		IETF Working groups TSVWG, NSIS, SIP and SIPPING

To:		Georges Sebek  <tsbsg13@itu.int> ,
		Keith Knightson <kgk@igs.net>

Cc:		Magnus Westerlund <magnus.westerlund@ericsson.com>,
                 Lars Eggert <lars.eggert@nokia.com>,
                 James Polk <jmpolk@cisco.com>,
                 Martin Stiemerling <stiemerling@netlab.nec.de>,
                 John Loughney <john.loughney@nokia.com>
                 Dean Willis <dean.willis@softarmor.com>,
                 Keith Drage <drage@alcatel-lucent.com>,
                 Mary Barnes <mary.barnes@nortel.com>,
                 Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>,
                 Scott Bradner <sob@harvard.edu>,
                 Kimberly King <ksking@mitre.org>,
                 Scott Brim <swb@employees.org>

Response
Contact:	TSV WG (tsv-chairs@tools.ietf.org)
		NSIS (nsis-chairs@tools.ietf.org)
		SIPPING (sipping-chairs@tools.ietf.org)
		SIP (sip-chairs@tools.ietf.org)
=09
Purpose: 	Response to Liaison Request

Deadline:	None

Title: Response to liaison request from ITU-T Study Group 13 work on=20
emergency telecommunications

This is a response to the liaison request made by Q3/13 regarding=20
emergency telecommunications (REF: NGN-GSI/DOC =E2=80=93 60 Rev.2). We wo=
uld=20
first like to apologize for failing to meet the deadline in responding.=20
We do hope that the late answer still can be of some use.

The liaison request was made to the following working groups (WG):=20
IEPREP, TSVWG, and NSIS. Since the request was sent the IEPREP WG has=20
concluded. In addition to the WGs from which information was requested,=20
we also think that work performed by the SIP and SIPPING WG may be of=20
relevance.

Pursuant to ITU-T Study Group 13 request for information on relevant and=20
related work in the IETF regarding emergency communications, we list=20
below RFCs and works in progress that we feel may be of interest to your=20
group.  Some of the completed work is Informational, and others are in=20
the category of Standards Track.

Q.3/13 also requested that we keep you informed of any developments in=20
regards to emergency telecommunications. In those regards we would like=20
to make you aware that IETF mailing list participation and document=20
information is free and open for anyone. Allowing participants in Q.3/13=20
to keep them selves informed of any developments. If Q.3/13 desire to=20
get further information about specific ongoing work, then please send a=20
liaison request to the responsible WG for those specific documents.

The following list is divided under the associated working groups from=20
which the work has been done within the IETF.


SIP Working Group

RFC 4411

Title:
    Extending the Session Initiation Protocol (SIP)
    Reason Header for Preemption Events

Abstract:
   This document proposes an IANA Registration extension to the Session
    Initiation Protocol (SIP) Reason Header to be included in a BYE
    Method Request as a result of a session preemption event, either at a
    user agent (UA), or somewhere in the network involving a
    reservation-based protocol such as the Resource ReSerVation Protocol
    (RSVP) or Next Steps in Signaling (NSIS).  This document does not
    attempt to address routers failing in the packet path; instead, it
    addresses a deliberate tear down of a flow between UAs, and informs
    the terminated UA(s) with an indication of what occurred.

RFC 4412

Title:
    Communications Resource Priority for the Session Initiation
    Protocol (SIP)

Abstract:
    This document defines two new Session Initiation Protocol (SIP)
    header fields for communicating resource priority, namely,
    "Resource-Priority" and "Accept-Resource-Priority".  The
    "Resource-Priority" header field can influence the behavior of SIP
    user agents (such as telephone gateways and IP telephones) and SIP
    proxies.  It does not directly influence the forwarding behavior of
    IP routers.



SIPPING Working Group

    The below two documents are proposed work items not yet accepted as
    IETF WG items but are likely of interest as they update RFC 4412.

Work In Progress:   draft-polk-sip-rph-in-responses-00

Title:
    Allowing SIP Resource Priority Header in SIP Responses

Abstract:
    The Session Initiation Protocol (SIP) Resource Priority Header
    (RPH), in its current form, is ignored in SIP responses.  This was a
    design choice during RFC 4412's development. This is now considered
    a bad design choice in certain scenarios.  This document corrects
    RFC 4412's communications model by optionally allowing a SIP server
    or user agent client to process the Resource-Priority Header in a
    response.

Work In Progress:   draft-polk-sip-rph-new-namespaces--00

Title:
    New Session Initiation Protocol Resource-Priority Header Namespaces
    for the Defense Information Systems Agency

Abstract:
    This document creates additional Session Initiation Protocol
    Resource-Priority header namespaces, to be IANA registered.  This
    document intends to update RFC 4412, as a Proposed Standard document
    if published by the RFC-Editor.

IEPREP Working Group

RFC 3487

Title:
    Requirements for Resource Priority Mechanisms for the
    Session Initiation Protocol (SIP)

Abstract:
    This document summarizes requirements for prioritizing access to
    circuit-switched network, end system and proxy resources for
    emergency preparedness communications using the Session Initiation
    Protocol (SIP).

RFC 3689

Title:
    General Requirements for
    Emergency Telecommunication Service (ETS)

Abstract:
    This document presents a list of general requirements in support of
    Emergency Telecommunications Service (ETS).  Solutions to these
    requirements are not presented in this document.  Additional
    requirements pertaining to specific applications, or types of
    applications, are to be specified in separate document(s).

RFC 3690

Title:
    IP Telephony Requirements for
    Emergency Telecommunication Service (ETS)

Abstract:
    This document presents a list of requirements in support of Emergency
    Telecommunications Service (ETS) within the context of IP telephony.
    It is an extension to the general requirements presented in RFC 3689.
    Solutions to these requirements are not presented in this document.

RFC 4190

Title:
    Framework for Supporting Emergency Telecommunications
    Service (ETS) in IP Telephony

Abstract:
    This document presents a framework for supporting authorized,
    emergency-related communication within the context of IP telephony.
    We present a series of objectives that reflect a general view of how
    authorized emergency service, in line with the Emergency
    Telecommunications Service (ETS), should be realized within today's
    IP architecture and service models.  From these objectives, we
    present a corresponding set of protocols and capabilities, which
    provide a more specific set of recommendations regarding existing
    IETF protocols.  Finally, we present two scenarios that act as
    guiding models for the objectives and functions listed in this
    document.  These models, coupled with an example of an existing
    service in the Public Switched Telephone Network (PSTN), contribute
    to a constrained solution space.

RFC 4375

Title:
    Emergency Telecommunications Services (ETS) Requirements
    for a Single Administrative Domain

Abstract:
    This document presents a list of requirements in support of Emergency
    Telecommunications Service (ETS) within a single administrative
    domain.  This document focuses on a specific set of administrative
    constraints and scope.  Solutions to these requirements are not
    presented in this document.


TSV Working Group

RFC 4542

Title:
    Implementing an Emergency Telecommunications Service (ETS)
    for Real-Time Services in the Internet Protocol Suite

Abstract:
    RFCs 3689 and 3690 detail requirements for an Emergency
    Telecommunications Service (ETS), of which an Internet Emergency
    Preparedness Service (IEPS) would be a part.  Some of these types of
    services require call preemption; others require call queuing or
    other mechanisms.  IEPS requires a Call Admission Control (CAC)
    procedure and a Per Hop Behavior (PHB) for the data that meet the
    needs of this architecture.  Such a CAC procedure and PHB is
    appropriate to any service that might use H.323 or SIP to set up
    real-time sessions.  The key requirement is to guarantee an elevated
    probability of call completion to an authorized user in time of
    crisis.

    This document primarily discusses supporting ETS in the context of
    the US Government and NATO, because it focuses on the Multi-Level
    Precedence and Preemption (MLPP) and Government Emergency
    Telecommunication Service (GETS) standards.  The architectures
    described here are applicable beyond these organizations.

Work In Progress:   draft-ietf-tsvwg-emergency-rsvp-03.txt

Title:
    Resource ReSerVation Protovol (RSVP) Extensions for
    Emergency Services

Abstract:
    An Emergency Telecommunications Service (ETS) requires the ability to
    provide an elevated probability of session establishment to an
    authorized user in times of network congestion (typically, during a
    crisis). When supported over the Internet Protocol suite, this may be
    facilitated through a network layer admission control solution, which
    supports prioritized access to resources (e.g., bandwidth). These
    resources may be explicitly set aside for emergency services, or they
    may be shared with other sessions.

   This document specifies RSVP extensions that can be used to support
    such an admission priority capability at the network layer. Note that
    these extensions represent one possible solution component in
    satisfying ETS requirements. Other solution components, or other
    solutions, are outside the scope of this document.

NSIS Working Group

Work In Progress: draft-ietf-nsis-qos-nslp-14.txt

Title:
    NSLP for Quality-of-Service Signaling

Abstract:
    This specification describes the NSIS Signaling Layer Protocol
    (NSLP) for signaling QoS reservations in the Internet.  It is in
    accordance with the framework and requirements developed in NSIS.
    Together with GIST, it provides functionality similar to RSVP and
    extends it.  The QoS NSLP is independent of the underlying QoS
    specification or architecture and provides support for different
    reservation models. It is simplified by the elimination of support
    for multicast flows. This specification explains the overall
    protocol approach, design decisions made and provides examples.  It
    specifies object, message  formats and processing rules.

--=20

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM/M
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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--=_alternative 0055890785257329_=--



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On Jul 30, 2007, at 8:36 PM, Qian Sun wrote:

> Hi, Dean
>
> Sorry for late response. Do you mean "advertisement" content always  
> is needed to 'render'? So a single content disposition  
> "advertisement" is enough.

Well, I'm wondering if advertisements have any other need than  
"render". If not, why not just use "render"?

If we define an "advertisement" content disposition, we might as well  
call it "ignore", in which case there's hardly any point in sending  
it to start with, right?

Although I must admit I'm intrigued by the prospect of getting  
advertisers to self-label stuff so that it can be discarded as  
efficiently as possible.

> In addition, some content body (e.g. Geolocation or advertisement)  
> may be inserted or modified by intermediary. Could we specify a  
> common way for indicating the source of inserted content body? For  
> example, define a new MIME header: Content-Insertedby. A B2BUA may  
> insert a content body including AD like this:
>
> Content-Type: text/plain
> Content-Disposition: advertisement
> Content-Insertedby: AD-Agent <sip: adagent@example.com>

I'm sure there has been discussion of this in geopriv, but I don't  
recall the details. I suspect there have been concerns about  
mismatches between the "source" and the "information".

--
Dean



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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Michael,

The semantics of a request to a sip: resource are different from a  
request to a sips: resource.  If there are no Contacts registered  
against a sips: resource, then the correct thing for the proxy to do  
is to send a 480.  This is the same thing the proxy would do if there  
were no Contacts registered against a sip: resource.

We are not talking about a ciphersuite downgrade here. Providing an  
error response code that encourages a client to downgrade is just  
plain irresponsible.

thanks,
-rohan


On Jul 27, 2007, at 9:39 AM, Michael Thomas wrote:
> Rohan Mahy wrote:
>> Michael,
>>
>> At issue here is what the default implementor is likely to do.   
>> With a new 4xx, the misguided but well-meaning implementor is  
>> likely to try to "helpfully" "repair" the error without thinking  
>> about or understanding the security context.
>>
>> Using a Warning code raises the bar significantly, but still  
>> allows automata to at least log what happened.
> As I said, a receiver is completely at liberty to prevent the  
> downgrade by not
> accepting the downgraded request.  Problem solved by those who  
> care. On
> the other hand I think it's pretty presumptuous that we "know" in  
> all cases
> whether it's wrong to "repair" the error. Paul brought up several  
> examples.
> Also: if the receiver allows non-SIPS requests why shouldn't you  
> infer that
> they want you to "repair" the request? It sure seems like this is a  
> misguided
> attempt at forcing ineffectual nannyware which is a pointless  
> enterprise for
> IETF protocols.
>
>       Mike



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From sip-bounces@ietf.org Tue Jul 31 12:15:56 2007
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Subject: RE: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
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Thanks. I'm OK with this.

=20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Tuesday, July 31, 2007 08:12
> To: Audet, Francois (SC100:3055)
> Cc: Cullen Jennings; Rohan Mahy; IETF SIP List;=20
> drage@alcatel-lucent.com; Robert Sparks;=20
> Jon.Peterson@neustar.com; Dean Willis
> Subject: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
>=20
> Fran=E7ois,
>=20
> I think the IANA paragraph you described is too restrictive. =20
> The text in RFC3261 section 20.43 seems perfectly fine with=20
> registering non-SDP errors.  I would allocate warn-code 380=20
> "No SIPS contacts registered".
>=20
> thanks,
> -rohan
>=20
> On Jul 30, 2007, at 3:23 PM, Francois Audet wrote:
>=20
> > Now that we have everybody exited about the prospect of using a=20
> > Warn-Code for "SIPS Not Allowed" and "SIP Required" with=20
> Response 480,=20
> > instead of using new response codes, here is a quote from 27.2/RFC=20
> > 3261.
> >
> >    Warning codes provide information supplemental to the=20
> status code=20
> > in
> >    SIP response messages when the failure of the transaction results
> >    from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.
> >
> > My reading of this is that Warn-Codes are ONLY usable for=20
> SDP errors.
> >
> > Doesn't this disqualify the idea of using a Warn-Code for=20
> SIP/SIPS URI=20
> > problems??????
> >
> > If so, aren't we back to 418/419, or 418+New header (Allow/=20
> Require),=20
> > or
> > 480+Response text?
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


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Subject: RE: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
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As Jame's has already remarked, we did discuss this in the context of =
location-conveyance.

I believe the SIP list conclusion at that point was that the RFC 3261 =
text was not preventing usage, but the IANA registry text did.

The Warning header registry was created by RFC 3261 in the days before =
formal IANA considerations sections in RFCs. As draft-ietf-sip-sips =
updates RFC 3261, it would seem to me that it is entirely appropriate =
for that document to update the IANA registry for Warning headers. By =
this I mean you need to update the registry definition itself, as well =
as add new values to it.

This is of course assuming that we decide that Warning headers are the =
way to go.

Regards

Keith

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Tuesday, July 31, 2007 4:12 PM
> To: Francois Audet
> Cc: Cullen Jennings; Rohan Mahy; IETF SIP List; DRAGE, Keith=20
> (Keith); Robert Sparks; Jon.Peterson@neustar.com; Dean Willis
> Subject: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
>=20
> Fran=E7ois,
>=20
> I think the IANA paragraph you described is too restrictive. =20
> The text in RFC3261 section 20.43 seems perfectly fine with=20
> registering non-SDP errors.  I would allocate warn-code 380=20
> "No SIPS contacts registered".
>=20
> thanks,
> -rohan
>=20
> On Jul 30, 2007, at 3:23 PM, Francois Audet wrote:
>=20
> > Now that we have everybody exited about the prospect of using a=20
> > Warn-Code for "SIPS Not Allowed" and "SIP Required" with=20
> Response 480,=20
> > instead of using new response codes, here is a quote from 27.2/RFC=20
> > 3261.
> >
> >    Warning codes provide information supplemental to the=20
> status code=20
> > in
> >    SIP response messages when the failure of the transaction results
> >    from a Session Description Protocol (SDP) (RFC 2327 [1]) problem.
> >
> > My reading of this is that Warn-Codes are ONLY usable for=20
> SDP errors.
> >
> > Doesn't this disqualify the idea of using a Warn-Code for=20
> SIP/SIPS URI=20
> > problems??????
> >
> > If so, aren't we back to 418/419, or 418+New header (Allow/=20
> Require),=20
> > or
> > 480+Response text?
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org Tue Jul 31 12:56:54 2007
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Subject: RE: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
Date: Tue, 31 Jul 2007 11:56:28 -0500
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Thread-Topic: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
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From: "Francois Audet" <audet@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>,
	"Rohan Mahy" <rohan@ekabal.com>, "James M. Polk" <jmpolk@cisco.com>,
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Looks like the Warning Codes approach is preferred.

I'm glad I'm now reading  =
http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-08=
.txt.

I almost fell off my chair when I saw the following two Error codes =
defined in
James' draft:

3.4.12 Warning code 720 Unsupported Schema - sip desired


   A Warning header with the code 720 "Unsupported Schema - sip
   desired" means the location dereferencer cannot dereference using
   the location-by-reference URI schema supplied because it does not
   support the necessary protocol to do this.  This Warning code means
   the location recipient can dereference the target's location using a
   sip-URI schema.  There can be more than one Warning code in a
   Warning header, indicated in this context the recipient can
   dereference using each schema protocol included in the Warning
   header.

   A typical reaction to receiving this warning code would be for the
   location sender to send a URI with the sip schema.

   Recommended warn-text: unsupported schema

   An example usage in a SIP 424 response:

   Warning: 720 alice.example.com "unsupported schema - sip desired"


3.4.12 Warning code 721 Unsupported Schema - sips desired


   A Warning header with the code 721 "Unsupported Schema - sips
   desired" means the location dereferencer cannot dereference using
   the location-by-reference URI schema supplied because it does not
   support the necessary protocol to do this.  This Warning code means
   the location recipient can dereference the target's location using a
   sips-URI schema.  There can be more than one Warning code in a
   Warning header, indicated in this context the recipient can
   dereference using each schema protocol included in the Warning
   header.

   Recommended warn-text: unsupported schema

   An example usage in a SIP 424 response:

   Warning: 721 alice.example.com "unsupported schema - sips desired"

As James pointed out, in -07, those were Warning headers, now they
are Geoprive headers.

If we are going to use 2 new Warning Headers in draft-ietf-sip-sips (say
380 and 381), wouldn't it make more sense if =
draft-ietf-sip-location-conveyance-09
use those values (with 424) instead of 720/721?

It seems to me that error 720-723 would be potentially applicable to =
more
than just Location. 700-711 however are specific to location, and should =
remain
as defined.

Any toughts?

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]=20
> Sent: Tuesday, July 31, 2007 09:20
> To: Rohan Mahy; Audet, Francois (SC100:3055)
> Cc: Cullen Jennings; IETF SIP List; Jon.Peterson@neustar.com;=20
> Robert Sparks; Dean Willis
> Subject: RE: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
>=20
> As Jame's has already remarked, we did discuss this in the=20
> context of location-conveyance.
>=20
> I believe the SIP list conclusion at that point was that the=20
> RFC 3261 text was not preventing usage, but the IANA registry=20
> text did.
>=20
> The Warning header registry was created by RFC 3261 in the=20
> days before formal IANA considerations sections in RFCs. As=20
> draft-ietf-sip-sips updates RFC 3261, it would seem to me=20
> that it is entirely appropriate for that document to update=20
> the IANA registry for Warning headers. By this I mean you=20
> need to update the registry definition itself, as well as add=20
> new values to it.
>=20
> This is of course assuming that we decide that Warning=20
> headers are the way to go.
>=20
> Regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan@ekabal.com]
> > Sent: Tuesday, July 31, 2007 4:12 PM
> > To: Francois Audet
> > Cc: Cullen Jennings; Rohan Mahy; IETF SIP List; DRAGE,=20
> Keith (Keith);=20
> > Robert Sparks; Jon.Peterson@neustar.com; Dean Willis
> > Subject: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
> >=20
> > Fran=E7ois,
> >=20
> > I think the IANA paragraph you described is too restrictive. =20
> > The text in RFC3261 section 20.43 seems perfectly fine with=20
> > registering non-SDP errors.  I would allocate warn-code 380=20
> "No SIPS=20
> > contacts registered".
> >=20
> > thanks,
> > -rohan
> >=20
> > On Jul 30, 2007, at 3:23 PM, Francois Audet wrote:
> >=20
> > > Now that we have everybody exited about the prospect of using a=20
> > > Warn-Code for "SIPS Not Allowed" and "SIP Required" with
> > Response 480,
> > > instead of using new response codes, here is a quote from=20
> 27.2/RFC=20
> > > 3261.
> > >
> > >    Warning codes provide information supplemental to the
> > status code
> > > in
> > >    SIP response messages when the failure of the=20
> transaction results
> > >    from a Session Description Protocol (SDP) (RFC 2327=20
> [1]) problem.
> > >
> > > My reading of this is that Warn-Codes are ONLY usable for
> > SDP errors.
> > >
> > > Doesn't this disqualify the idea of using a Warn-Code for
> > SIP/SIPS URI
> > > problems??????
> > >
> > > If so, aren't we back to 418/419, or 418+New header (Allow/
> > Require),
> > > or
> > > 480+Response text?
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Francois

I don't believe it is a glitch so much as it is an initial range, 
which can be expanded with a PS RFC, especially one with an update to 
3261 (which you are doing).

I was using 7XX (assigning 701 through 739) - though I pulled the 
Warning codes from the conveyance draft, therefore I am no longer 
using these codes (since I created a new SIP header 
(Geolocation-Error) to accomplish the same task as the Warning codes 
plus add new information with header parameters for the new header).

My point was that it is possible to go outside the 3XX range for 
SIP... you just have to define it.

BTW - I am ok with the 480+Warning Code, but am cautious about the 
Warning header (with its 'why' information) being stripped along the 
way back to the UAC.

Ultimately, do we want zero communications between this UAC and UAS 
(because the UAC can always get a 480 without additional information 
letting it know what was really wrong with the request), or do we 
want to send the UAC something it can understand about this scenario?

At 08:49 AM 7/31/2007, Francois Audet wrote:
>Ok, that's fine with me. It's just a glitch in 3261 then. I'd rather we
>stick with the Warn-Codes approach.
>
>Did you assign a purpose for 7XX-series warn-codes yet? I'm wondering
>what
>range should be used for this purpose.
>
>James, I didn't see your codes on IANA's web site. What values are you
>using?
>I don't want to overlap.
>
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Monday, July 30, 2007 21:44
> > To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat
> > Paul; Mahy Rohan
> > Cc: IETF SIP List; Jon.Peterson@neustar.com
> > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >
> > At 10:54 AM 7/30/2007, Francois Audet wrote:
> > >RFC 3261, p. 181:
> > >
> > >         "The first digit  of warning codes beginning with "3"
> > >indicates warnings
> > >         specific to SIP."
> > >
> > >Then it explains within the 3XX space, how they are used.
> >
> > btw -- for location-conveyance-07, I created 15 new Warning codes (#s
> > 701 through 712, then 720, 721, and 722).  All this at the
> > suggestion of Henning (who knows something about 3261 text...  ;-)
> >
> > btw II - I have since replaced the Warning header indication
> > with a new Geolocation-Error header, but this was so I could
> > stuff more into the new header than the Warning header would allow.
> >
> > In all this, I did not get one person claiming Warning codes
> > were limited to 3XX.  Henning merely said the "first batch of
> > codes were in the 300s, and that there was nothing limiting a
> > new ID from defining new codes in any other number range"...
> >
> >
> > > > -----Original Message-----
> > > > From: James M. Polk [mailto:jmpolk@cisco.com]
> > > > Sent: Saturday, July 28, 2007 18:25
> > > > To: Audet, Francois (SC100:3055); Alan Hawrylyshen; Kyzivat Paul;
> > > > Mahy Rohan
> > > > Cc: IETF SIP List; Jon.Peterson@neustar.com
> > > > Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > > >
> > > > A 399 Miscellaneous?  That doesn't make sense...
> > > >
> > > > Warning codes are not limited to 3XX, so if a new Warning code is
> > > > what Rohan wants to convey more information in the
> > response (which I
> > > > believe should be done one way or another)
> > > > - it can be almost any value other than 300-379 or 399 IMO.
> > > >
> > > > At 10:32 AM 7/27/2007, Francois Audet wrote:
> > > > >On the Warning header...
> > > > >
> > > > >Values 300-379 are reserved for SDP-related stuff.
> > > > >
> > > > >Values 380-389 are unassiged (but there is no text on it).
> > > > >
> > > > >Values 390-398 are unassigned (but don't fall into the
> > SDP related
> > > > >stuff.
> > > > >
> > > > >399 means Miscellaneous and the text can be provided to the user.
> > > > >It further says that Automata MUST NOT take action based on this.
> > > > >
> > > > >Rohan, are you advocating using 399, or defining a new code from
> > > > >the
> > > > >390-398 space?
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Alan Hawrylyshen [mailto:a.hawrylyshen@gmail.com]
> > > > On Behalf Of
> > > > > > Alan Hawrylyshen
> > > > > > Sent: Friday, July 27, 2007 08:27
> > > > > > To: Kyzivat Paul; Mahy Rohan
> > > > > > Cc: Audet, Francois (SC100:3055); IETF SIP List;
> > > > > > Jon.Peterson@neustar.com
> > > > > > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> > > > > >
> > > > > >
> > > > > > On 27-Jul-2007, at 10:06 , Paul Kyzivat wrote:
> > > > > >
> > > > > > > I understand that if the user entered a sips URI then it
> > > > > > should be the
> > > > > > > user that must decide to downgrade. But if the user didn't
> > > > > > > know whether to use sip or sips in the first place,
> > and the UA
> > > > > > decides to
> > > > > > > try sips first then I see no problem in the UA
> > having a policy
> > > > > > > that causes it to downgrade.
> > > > > >
> > > > > > I was under the impression (based on meeting
> > discussion) that :
> > > > > > 1 - the downgrade was undesirable because it reveals
> > > > > > (possibly) information about the targeted party in the clear,
> > > > > > and;
> > > > > > 2 - The 480 with a Warning header was an option to provide
> > > > > > automata- friendly indication of the failure reason.
> > > > > >
> > > > > > Alan Hawrylyshen
> > > > > >
> > > > > > a l a n a t p o l y p h a s e d o t c a
> > > > > >
> > > > > >
> > > > > >
> > > > >
> > > > >
> > > > >_______________________________________________
> > > > >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > >This list is for NEW development of the core SIP Protocol Use
> > > > >sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > > >sipping@ietf.org for new developments on the application of sip
> > > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Tue, 31 Jul 2007 12:51:17 -0500
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=20
> Ultimately, do we want zero communications between this UAC=20
> and UAS (because the UAC can always get a 480 without=20
> additional information letting it know what was really wrong=20
> with the request), or do we want to send the UAC something it=20
> can understand about this scenario?

Yes, we want the default to be a "hard failure" that looks like if the
target was not corrently registered.

That's the reasoning behind the 480.


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From sip-bounces@ietf.org Tue Jul 31 17:25:49 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Tue, 31 Jul 2007 17:25:08 -0400
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Yeah but a 480 implies the resource is not available, whereas in reality the
resource *is* available using a different scheme.  I guess one question is
if the sips scheme literally means the UAS is a different "resource".  If
so, then registering a sips contact only should not implicitly make the UAS
available for routing plain sip.  The draft currently says they represent
the *same* resource.

I'm not convinced by the downgrade argument.  The far-end can just as easily
send a 3xx redirecting the UAC to the sip contact.  Per this new draft the
proxies must not recurse on that, and the UAC should inform the user before
doing so.  But I think a new 4xx response code would be much more likely to
make that a reality, as any vendor implementing support for recursing on a
418 would have read the draft and seen where it says "inform the user".  If
they didn't read the draft, they'd fail the request since they got an
unknown 4xx response. (compare that to a 3xx, which is known/usable without
reading this draft)  Am I missing the gist of the concern?

-hadriel


> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]
> Sent: Tuesday, July 31, 2007 12:04 PM
> To: Michael Thomas
> Cc: Rohan Mahy; IETF SIP List; Francois Audet; Jon.Peterson@neustar.com;
> Paul Kyzivat
> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> 
> Michael,
> 
> The semantics of a request to a sip: resource are different from a
> request to a sips: resource.  If there are no Contacts registered
> against a sips: resource, then the correct thing for the proxy to do
> is to send a 480.  This is the same thing the proxy would do if there
> were no Contacts registered against a sip: resource.
> 
> We are not talking about a ciphersuite downgrade here. Providing an
> error response code that encourages a client to downgrade is just
> plain irresponsible.
> 
> thanks,
> -rohan
> 
> 
> On Jul 27, 2007, at 9:39 AM, Michael Thomas wrote:
> > Rohan Mahy wrote:
> >> Michael,
> >>
> >> At issue here is what the default implementor is likely to do.
> >> With a new 4xx, the misguided but well-meaning implementor is
> >> likely to try to "helpfully" "repair" the error without thinking
> >> about or understanding the security context.
> >>
> >> Using a Warning code raises the bar significantly, but still
> >> allows automata to at least log what happened.
> > As I said, a receiver is completely at liberty to prevent the
> > downgrade by not
> > accepting the downgraded request.  Problem solved by those who
> > care. On
> > the other hand I think it's pretty presumptuous that we "know" in
> > all cases
> > whether it's wrong to "repair" the error. Paul brought up several
> > examples.
> > Also: if the receiver allows non-SIPS requests why shouldn't you
> > infer that
> > they want you to "repair" the request? It sure seems like this is a
> > misguided
> > attempt at forcing ineffectual nannyware which is a pointless
> > enterprise for
> > IETF protocols.
> >
> >       Mike
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From sip-bounces@ietf.org Tue Jul 31 17:51:07 2007
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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Tue, 31 Jul 2007 16:50:39 -0500
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480 means that the user identified by the Request-URI is recognized, but
there is currently no valid forwarding location for that user. In this
case,
no SIPS contacts.

What many people are missing in the 480 vs. 418 argument is the
following:

- Existing implementations will treat new 4XX error codes as 400,
  which basically means "the protocol message is broken".

- Existing implements will treat 480 with new Warning just like
  480 witn no warning, i.e., as "Temporarily unavailable".

There is a huge difference between the 2. 400 will lead you to believe
that there is something wrong with the equipment (i.e., you may file a
complaint with the manufacturer). Very bad.

480 just means the other end is not available at that address. Good.=20
Everybody is happy.

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: Tuesday, July 31, 2007 14:25
> To: 'Rohan Mahy'; 'Michael Thomas'
> Cc: 'IETF SIP List'; Audet, Francois (SC100:3055);=20
> Jon.Peterson@neustar.com; 'Paul Kyzivat'
> Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>=20
> Yeah but a 480 implies the resource is not available, whereas=20
> in reality the resource *is* available using a different=20
> scheme.  I guess one question is if the sips scheme literally=20
> means the UAS is a different "resource".  If so, then=20
> registering a sips contact only should not implicitly make=20
> the UAS available for routing plain sip.  The draft currently=20
> says they represent the *same* resource.
>=20
> I'm not convinced by the downgrade argument.  The far-end can=20
> just as easily send a 3xx redirecting the UAC to the sip=20
> contact.  Per this new draft the proxies must not recurse on=20
> that, and the UAC should inform the user before doing so. =20
> But I think a new 4xx response code would be much more likely=20
> to make that a reality, as any vendor implementing support=20
> for recursing on a
> 418 would have read the draft and seen where it says "inform=20
> the user".  If they didn't read the draft, they'd fail the=20
> request since they got an unknown 4xx response. (compare that=20
> to a 3xx, which is known/usable without reading this draft) =20
> Am I missing the gist of the concern?
>=20
> -hadriel
>=20
>=20
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan@ekabal.com]
> > Sent: Tuesday, July 31, 2007 12:04 PM
> > To: Michael Thomas
> > Cc: Rohan Mahy; IETF SIP List; Francois Audet;=20
> > Jon.Peterson@neustar.com; Paul Kyzivat
> > Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
> >=20
> > Michael,
> >=20
> > The semantics of a request to a sip: resource are different from a=20
> > request to a sips: resource.  If there are no Contacts registered=20
> > against a sips: resource, then the correct thing for the=20
> proxy to do=20
> > is to send a 480.  This is the same thing the proxy would=20
> do if there=20
> > were no Contacts registered against a sip: resource.
> >=20
> > We are not talking about a ciphersuite downgrade here. Providing an=20
> > error response code that encourages a client to downgrade is just=20
> > plain irresponsible.
> >=20
> > thanks,
> > -rohan
> >=20
> >=20
> > On Jul 27, 2007, at 9:39 AM, Michael Thomas wrote:
> > > Rohan Mahy wrote:
> > >> Michael,
> > >>
> > >> At issue here is what the default implementor is likely to do.
> > >> With a new 4xx, the misguided but well-meaning implementor is=20
> > >> likely to try to "helpfully" "repair" the error without thinking=20
> > >> about or understanding the security context.
> > >>
> > >> Using a Warning code raises the bar significantly, but=20
> still allows=20
> > >> automata to at least log what happened.
> > > As I said, a receiver is completely at liberty to prevent the=20
> > > downgrade by not accepting the downgraded request. =20
> Problem solved=20
> > > by those who care. On the other hand I think it's pretty=20
> > > presumptuous that we "know" in all cases whether it's wrong to=20
> > > "repair" the error. Paul brought up several examples.
> > > Also: if the receiver allows non-SIPS requests why shouldn't you=20
> > > infer that they want you to "repair" the request? It sure=20
> seems like=20
> > > this is a misguided attempt at forcing ineffectual=20
> nannyware which=20
> > > is a pointless enterprise for IETF protocols.
> > >
> > >       Mike
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20


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Subject: RE: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
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Hi all,

It seems from the mailing list discussion that the preferred approach is
to
use the 480 (Temporarily Unavailable) response code, with a Warning
header field.

The Warning header could use one of the following 2 newly defined
warn-codes:

   380  SIPS Not Allowed: The UAS or proxy can process the request
        because the SIPS scheme is not allowed (e.g., because there are
        currently no registered SIPS Contacts).

   381  SIPS Required: The UAS or proxy cannot process the request
        because the SIPS scheme is required.

If you cannot live with this approach, please voice your opinion.

Thanks.


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From sip-bounces@ietf.org Tue Jul 31 19:36:09 2007
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Mary,

thanks much for reviewing this document. I've fixed all the nits you 
have identified. Thanks much!

Volker



Mary Barnes wrote:
> Draft:  draft-ietf-sip-session-policy-framework-01
> Reviewer: Mary Barnes
> Review Date: 8 June 2007
> Review Deadline: 11 June 2007
> Status: WGLC
> 
> Summary: This draft is basically ready for publication, but has nits 
> that should be fixed before publication. 
> 
> Comments:
> ----------
> I reviewed the diff for this document as compared to the version 
> I previously reviewed:
> http://www.softarmor.com/sipping/process/wg-review/reviews/draft-ietf-si
> pping-session-policy-framework-01-barnes.txt
> 
> And, I believe all my previous concerns have been addressed.  There are
> a 
> few nits that I noticed that should be fixed.
> 
> 
> Nits:
> ------
> - Header: Need to add "Intended Status:" (i.e., category="std" in the
> <rfc> field) 
> 
> - Section 4.4.1:
> -  2nd paragraph, 5th sentence: insert "the" between "using" and
> "mechanism" in the following sentence:
>    "If the UAC decides to continue, it MUST use the policy server 
>    URI to contact the policy server using the mechanism defined in 
>    Section 4.5."
> 
> - 2nd paragraph, 8th sentence: "resend" -> "resent" in the following
> sentence:
>    "If no changes are required by policies or no policies have
>    been received, the request can be resend without any policy-induced
>    changes."


-- 
Volker Hilt                      Bell Labs / Alcatel-Lucent
phone: +1 732 888 7206           791 Holmdel-Keyport Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733



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From sip-bounces@ietf.org Tue Jul 31 19:36:09 2007
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Roni,

thanks much for reviewing this document. Comments inline.

> 1. The draft explains describes two mechanisms the session 
> independent and the session dependent policies. I did not see what
> the relationship between them if any is. I think that at least the
> draft should say that they can be implemented independently if there
> are no other recommendations.
> 
I've added two sentences to the end of the introduction clarifying this. 
If also added a sentence to the end of the Session-independent policy 
section clarifying what to do when both mechanisms are used.

> 2. Section 3.2 describes how a UA discovers the policy servers for
> session independent policies based on the configuration framework
> ua-profile.
> 
> "ua-profile" event package [4] provides a mechanism to discover
>    policy servers in the local network and the home domain.  The "local-
>    network" profile-type enables a UA to discover a policy server in the
>    local domain.  The "user" profile type enables the discovery of a
>    policy server in the home domain.  A UA compliant to this
>    specification SHOULD attempt to discover and subscribe to the policy
>    servers in these two domains"
> 
> The configuration framework does not specify the profile data which
> is out of scope.

The profile type is not the profile data itself. The profile type 
effectively defines how the UA subscribes to the policy server in a 
given domain (e.g. in the local or home domain).

> So what is the meaning of this section - is this one way to find the 
> policy server 

Yes.

> or do you suggest that if you want to have session 
> independent policies you should have the policy servers defined in
> the profile data but then how does a UA identify it in the profile
> data, is it by name?
>
No. The idea is that the UA uses the procedures defined in the config
framework to construct the URI of these servers.

> I think this needs more clarification.
> 
I've revised the above paragraph to make this more clear.

> 3. In section 4.2
> " Endpoints set up a separate policy channel to each policy
>    server and can specifically decide which information they want to
>    disclose to which policy server"
> How does this work with the trust model between the UA and the proxy.
> The UA claimed that it got the policy from the proxy but from this 
> sentence you can assume that the UA got the policy only to what it 
> asked. I think that the UA need to include in document sent to the
> proxy all the information it can map from SDP.
> 
True. The UA can disclose what is requested by a specific policy server 
(e.g. offer only or offer/answer). I'll change this sentence accordingly.

> 4. In section 4.4.2 what will be the proxy behavior if it sends the
> Invite with the policy server URI to an endpoint who do not support
> session policy. The EP can ignore the request and respond 200 that will
> not include the support for the session policy. What will the proxy do
> or is it left to local implementation.
> 
The task of the proxy is to convey the policy server URI to the UAS. The 
enforcement that the UAS actually uses the URI is outside the scope here 
and could happen in the proxy or on the network level (i.e. it is left 
to local implementation).

Nevertheless, I think a proxy should be able to know whether the policy 
server URI was understood by the UAS (e.g. because it may want to do SDP 
modification if it was not understood). To enable this, the UAS now 
includes a Supported header into responses that contained a 
Policy-Contact header.

> 5. In section 4.4.6 in the note
> 
> " If, however, a policy server cannot respond with
>       a policy right away, it may return a policy that temporarily
>       denies the session and update this policy as the actual policy
>       decision becomes available"
> 
> I am not sure how does that work, what will the UA do in this case.
> Will it continue with the call without any media sessions?
> 
No, I don't think that would be terribly useful. If the policy server 
does not return the admittance decision right away, a UA can decide to 
wait, maybe informing the user with something like "All resources are 
current busy - Retrying". Alternatively, it could terminate the call 
attempt (maybe after a short time).
> 
> In section 4.4.3 third line should be field and not filed 
> 
Fixed.

Thanks much!

Volker








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At 04:58 PM 7/31/2007, Francois Audet wrote:
>Hi all,
>
>It seems from the mailing list discussion that the preferred approach is
>to
>use the 480 (Temporarily Unavailable) response code, with a Warning
>header field.
>
>The Warning header could use one of the following 2 newly defined
>warn-codes:
>
>    380  SIPS Not Allowed: The UAS or proxy can

s/can/cannot

>process the request
>         because the SIPS scheme is not allowed (e.g., because there are
>         currently no registered SIPS Contacts).
>
>    381  SIPS Required: The UAS or proxy cannot process the request
>         because the SIPS scheme is required.
>
>If you cannot live with this approach, please voice your opinion.
>
>Thanks.
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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Date: Wed, 1 Aug 2007 10:49:27 +1000
From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: "Francois Audet" <audet@nortel.com>
Subject: Re: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
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Yes, it makes more sense if draft-ietf-sip-location-conveyance-09 use 380
and 381 values (with 424) instead of 720/721.

Hisham


On 01/08/07, Francois Audet <audet@nortel.com> wrote:
>
> Looks like the Warning Codes approach is preferred.
>
> I'm glad I'm now reading
> http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-08=
.txt
> .
>
> I almost fell off my chair when I saw the following two Error codes
> defined in
> James' draft:
>
> 3.4.12 Warning code 720 Unsupported Schema - sip desired
>
>
>   A Warning header with the code 720 "Unsupported Schema - sip
>   desired" means the location dereferencer cannot dereference using
>   the location-by-reference URI schema supplied because it does not
>   support the necessary protocol to do this.  This Warning code means
>   the location recipient can dereference the target's location using a
>   sip-URI schema.  There can be more than one Warning code in a
>   Warning header, indicated in this context the recipient can
>   dereference using each schema protocol included in the Warning
>   header.
>
>   A typical reaction to receiving this warning code would be for the
>   location sender to send a URI with the sip schema.
>
>   Recommended warn-text: unsupported schema
>
>   An example usage in a SIP 424 response:
>
>   Warning: 720 alice.example.com "unsupported schema - sip desired"
>
>
> 3.4.12 Warning code 721 Unsupported Schema - sips desired
>
>
>   A Warning header with the code 721 "Unsupported Schema - sips
>   desired" means the location dereferencer cannot dereference using
>   the location-by-reference URI schema supplied because it does not
>   support the necessary protocol to do this.  This Warning code means
>   the location recipient can dereference the target's location using a
>   sips-URI schema.  There can be more than one Warning code in a
>   Warning header, indicated in this context the recipient can
>   dereference using each schema protocol included in the Warning
>   header.
>
>   Recommended warn-text: unsupported schema
>
>   An example usage in a SIP 424 response:
>
>   Warning: 721 alice.example.com "unsupported schema - sips desired"
>
> As James pointed out, in -07, those were Warning headers, now they
> are Geoprive headers.
>
> If we are going to use 2 new Warning Headers in draft-ietf-sip-sips (say
> 380 and 381), wouldn't it make more sense if
> draft-ietf-sip-location-conveyance-09
> use those values (with 424) instead of 720/721?
>
> It seems to me that error 720-723 would be potentially applicable to more
> than just Location. 700-711 however are specific to location, and should
> remain
> as defined.
>
> Any toughts?
>
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> > Sent: Tuesday, July 31, 2007 09:20
> > To: Rohan Mahy; Audet, Francois (SC100:3055)
> > Cc: Cullen Jennings; IETF SIP List; Jon.Peterson@neustar.com;
> > Robert Sparks; Dean Willis
> > Subject: RE: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
> >
> > As Jame's has already remarked, we did discuss this in the
> > context of location-conveyance.
> >
> > I believe the SIP list conclusion at that point was that the
> > RFC 3261 text was not preventing usage, but the IANA registry
> > text did.
> >
> > The Warning header registry was created by RFC 3261 in the
> > days before formal IANA considerations sections in RFCs. As
> > draft-ietf-sip-sips updates RFC 3261, it would seem to me
> > that it is entirely appropriate for that document to update
> > the IANA registry for Warning headers. By this I mean you
> > need to update the registry definition itself, as well as add
> > new values to it.
> >
> > This is of course assuming that we decide that Warning
> > headers are the way to go.
> >
> > Regards
> >
> > Keith
> >
> > > -----Original Message-----
> > > From: Rohan Mahy [mailto:rohan@ekabal.com]
> > > Sent: Tuesday, July 31, 2007 4:12 PM
> > > To: Francois Audet
> > > Cc: Cullen Jennings; Rohan Mahy; IETF SIP List; DRAGE,
> > Keith (Keith);
> > > Robert Sparks; Jon.Peterson@neustar.com; Dean Willis
> > > Subject: [Sip] Re: Warn-Codes and draft-ietf-sip-sips
> > >
> > > Fran=E7ois,
> > >
> > > I think the IANA paragraph you described is too restrictive.
> > > The text in RFC3261 section 20.43 seems perfectly fine with
> > > registering non-SDP errors.  I would allocate warn-code 380
> > "No SIPS
> > > contacts registered".
> > >
> > > thanks,
> > > -rohan
> > >
> > > On Jul 30, 2007, at 3:23 PM, Francois Audet wrote:
> > >
> > > > Now that we have everybody exited about the prospect of using a
> > > > Warn-Code for "SIPS Not Allowed" and "SIP Required" with
> > > Response 480,
> > > > instead of using new response codes, here is a quote from
> > 27.2/RFC
> > > > 3261.
> > > >
> > > >    Warning codes provide information supplemental to the
> > > status code
> > > > in
> > > >    SIP response messages when the failure of the
> > transaction results
> > > >    from a Session Description Protocol (SDP) (RFC 2327
> > [1]) problem.
> > > >
> > > > My reading of this is that Warn-Codes are ONLY usable for
> > > SDP errors.
> > > >
> > > > Doesn't this disqualify the idea of using a Warn-Code for
> > > SIP/SIPS URI
> > > > problems??????
> > > >
> > > > If so, aren't we back to 418/419, or 418+New header (Allow/
> > > Require),
> > > > or
> > > > 480+Response text?
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use
> > sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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<div>Yes, it makes more sense if draft-ietf-sip-location-conveyance-09 use =
380 and 381 values (with 424) instead of 720/721.</div>
<div>&nbsp;</div>
<div>Hisham<br><br>&nbsp;</div>
<div><span class=3D"gmail_quote">On 01/08/07, <b class=3D"gmail_sendername"=
>Francois Audet</b> &lt;<a href=3D"mailto:audet@nortel.com">audet@nortel.co=
m</a>&gt; wrote:</span>
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Looks like the Warning Codes app=
roach is preferred.<br><br>I&#39;m glad I&#39;m now reading&nbsp;&nbsp;<a h=
ref=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyan=
ce-08.txt">
http://www.ietf.org/internet-drafts/draft-ietf-sip-location-conveyance-08.t=
xt</a>.<br><br>I almost fell off my chair when I saw the following two Erro=
r codes defined in<br>James&#39; draft:<br><br>3.4.12 Warning code 720 Unsu=
pported Schema - sip desired
<br><br><br>&nbsp;&nbsp;A Warning header with the code 720 &quot;Unsupporte=
d Schema - sip<br>&nbsp;&nbsp;desired&quot; means the location dereferencer=
 cannot dereference using<br>&nbsp;&nbsp;the location-by-reference URI sche=
ma supplied because it does not
<br>&nbsp;&nbsp;support the necessary protocol to do this.&nbsp;&nbsp;This =
Warning code means<br>&nbsp;&nbsp;the location recipient can dereference th=
e target&#39;s location using a<br>&nbsp;&nbsp;sip-URI schema.&nbsp;&nbsp;T=
here can be more than one Warning code in a<br>
&nbsp;&nbsp;Warning header, indicated in this context the recipient can<br>=
&nbsp;&nbsp;dereference using each schema protocol included in the Warning<=
br>&nbsp;&nbsp;header.<br><br>&nbsp;&nbsp;A typical reaction to receiving t=
his warning code would be for the<br>
&nbsp;&nbsp;location sender to send a URI with the sip schema.<br><br>&nbsp=
;&nbsp;Recommended warn-text: unsupported schema<br><br>&nbsp;&nbsp;An exam=
ple usage in a SIP 424 response:<br><br>&nbsp;&nbsp;Warning: 720 <a href=3D=
"http://alice.example.com">alice.example.com
</a> &quot;unsupported schema - sip desired&quot;<br><br><br>3.4.12 Warning=
 code 721 Unsupported Schema - sips desired<br><br><br>&nbsp;&nbsp;A Warnin=
g header with the code 721 &quot;Unsupported Schema - sips<br>&nbsp;&nbsp;d=
esired&quot; means the location dereferencer cannot dereference using
<br>&nbsp;&nbsp;the location-by-reference URI schema supplied because it do=
es not<br>&nbsp;&nbsp;support the necessary protocol to do this.&nbsp;&nbsp=
;This Warning code means<br>&nbsp;&nbsp;the location recipient can derefere=
nce the target&#39;s location using a
<br>&nbsp;&nbsp;sips-URI schema.&nbsp;&nbsp;There can be more than one Warn=
ing code in a<br>&nbsp;&nbsp;Warning header, indicated in this context the =
recipient can<br>&nbsp;&nbsp;dereference using each schema protocol include=
d in the Warning<br>&nbsp;&nbsp;header.<br><br>
&nbsp;&nbsp;Recommended warn-text: unsupported schema<br><br>&nbsp;&nbsp;An=
 example usage in a SIP 424 response:<br><br>&nbsp;&nbsp;Warning: 721 <a hr=
ef=3D"http://alice.example.com">alice.example.com</a> &quot;unsupported sch=
ema - sips desired&quot;<br>
<br>As James pointed out, in -07, those were Warning headers, now they<br>a=
re Geoprive headers.<br><br>If we are going to use 2 new Warning Headers in=
 draft-ietf-sip-sips (say<br>380 and 381), wouldn&#39;t it make more sense =
if draft-ietf-sip-location-conveyance-09
<br>use those values (with 424) instead of 720/721?<br><br>It seems to me t=
hat error 720-723 would be potentially applicable to more<br>than just Loca=
tion. 700-711 however are specific to location, and should remain<br>as def=
ined.
<br><br>Any toughts?<br><br>&gt; -----Original Message-----<br>&gt; From: D=
RAGE, Keith (Keith) [mailto:<a href=3D"mailto:drage@alcatel-lucent.com">dra=
ge@alcatel-lucent.com</a>]<br>&gt; Sent: Tuesday, July 31, 2007 09:20<br>
&gt; To: Rohan Mahy; Audet, Francois (SC100:3055)<br>&gt; Cc: Cullen Jennin=
gs; IETF SIP List; <a href=3D"mailto:Jon.Peterson@neustar.com">Jon.Peterson=
@neustar.com</a>;<br>&gt; Robert Sparks; Dean Willis<br>&gt; Subject: RE: [=
Sip] Re: Warn-Codes and draft-ietf-sip-sips
<br>&gt;<br>&gt; As Jame&#39;s has already remarked, we did discuss this in=
 the<br>&gt; context of location-conveyance.<br>&gt;<br>&gt; I believe the =
SIP list conclusion at that point was that the<br>&gt; RFC 3261 text was no=
t preventing usage, but the IANA registry
<br>&gt; text did.<br>&gt;<br>&gt; The Warning header registry was created =
by RFC 3261 in the<br>&gt; days before formal IANA considerations sections =
in RFCs. As<br>&gt; draft-ietf-sip-sips updates RFC 3261, it would seem to =
me
<br>&gt; that it is entirely appropriate for that document to update<br>&gt=
; the IANA registry for Warning headers. By this I mean you<br>&gt; need to=
 update the registry definition itself, as well as add<br>&gt; new values t=
o it.
<br>&gt;<br>&gt; This is of course assuming that we decide that Warning<br>=
&gt; headers are the way to go.<br>&gt;<br>&gt; Regards<br>&gt;<br>&gt; Kei=
th<br>&gt;<br>&gt; &gt; -----Original Message-----<br>&gt; &gt; From: Rohan=
 Mahy [mailto:
<a href=3D"mailto:rohan@ekabal.com">rohan@ekabal.com</a>]<br>&gt; &gt; Sent=
: Tuesday, July 31, 2007 4:12 PM<br>&gt; &gt; To: Francois Audet<br>&gt; &g=
t; Cc: Cullen Jennings; Rohan Mahy; IETF SIP List; DRAGE,<br>&gt; Keith (Ke=
ith);
<br>&gt; &gt; Robert Sparks; <a href=3D"mailto:Jon.Peterson@neustar.com">Jo=
n.Peterson@neustar.com</a>; Dean Willis<br>&gt; &gt; Subject: [Sip] Re: War=
n-Codes and draft-ietf-sip-sips<br>&gt; &gt;<br>&gt; &gt; Fran=E7ois,<br>&g=
t; &gt;
<br>&gt; &gt; I think the IANA paragraph you described is too restrictive.<=
br>&gt; &gt; The text in RFC3261 section 20.43 seems perfectly fine with<br=
>&gt; &gt; registering non-SDP errors.&nbsp;&nbsp;I would allocate warn-cod=
e 380<br>
&gt; &quot;No SIPS<br>&gt; &gt; contacts registered&quot;.<br>&gt; &gt;<br>=
&gt; &gt; thanks,<br>&gt; &gt; -rohan<br>&gt; &gt;<br>&gt; &gt; On Jul 30, =
2007, at 3:23 PM, Francois Audet wrote:<br>&gt; &gt;<br>&gt; &gt; &gt; Now =
that we have everybody exited about the prospect of using a
<br>&gt; &gt; &gt; Warn-Code for &quot;SIPS Not Allowed&quot; and &quot;SIP=
 Required&quot; with<br>&gt; &gt; Response 480,<br>&gt; &gt; &gt; instead o=
f using new response codes, here is a quote from<br>&gt; 27.2/RFC<br>&gt; &=
gt; &gt; 3261.
<br>&gt; &gt; &gt;<br>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;Warning codes p=
rovide information supplemental to the<br>&gt; &gt; status code<br>&gt; &gt=
; &gt; in<br>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;SIP response messages wh=
en the failure of the<br>&gt; transaction results
<br>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;from a Session Description Protoc=
ol (SDP) (RFC 2327<br>&gt; [1]) problem.<br>&gt; &gt; &gt;<br>&gt; &gt; &gt=
; My reading of this is that Warn-Codes are ONLY usable for<br>&gt; &gt; SD=
P errors.<br>&gt; &gt; &gt;
<br>&gt; &gt; &gt; Doesn&#39;t this disqualify the idea of using a Warn-Cod=
e for<br>&gt; &gt; SIP/SIPS URI<br>&gt; &gt; &gt; problems??????<br>&gt; &g=
t; &gt;<br>&gt; &gt; &gt; If so, aren&#39;t we back to 418/419, or 418+New =
header (Allow/
<br>&gt; &gt; Require),<br>&gt; &gt; &gt; or<br>&gt; &gt; &gt; 480+Response=
 text?<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt; ________________=
_______________________________<br>&gt; &gt; Sip mailing list&nbsp;&nbsp;<a=
 href=3D"https://www1.ietf.org/mailman/listinfo/sip">
https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt; &gt; This list is fo=
r NEW development of the core SIP Protocol Use<br>&gt; &gt; <a href=3D"mail=
to:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a>
 for questions on current sip Use<br>&gt; &gt; <a href=3D"mailto:sipping@ie=
tf.org">sipping@ietf.org</a> for new developments on the application of sip=
<br>&gt; &gt;<br>&gt;<br>&gt;<br>&gt; _____________________________________=
__________
<br>&gt; Sip mailing list&nbsp;&nbsp;<a href=3D"https://www1.ietf.org/mailm=
an/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</a><br>&gt; Thi=
s list is for NEW development of the core SIP Protocol Use<br>&gt; <a href=
=3D"mailto:sip-implementors@cs.columbia.edu">
sip-implementors@cs.columbia.edu</a> for questions on current sip<br>&gt; U=
se <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new develop=
ments on the application of sip<br>&gt;<br><br><br>________________________=
_______________________
<br>Sip mailing list&nbsp;&nbsp;<a href=3D"https://www1.ietf.org/mailman/li=
stinfo/sip">https://www1.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-i=
mplementors@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 developments=
 on the application of sip<br></blockquote></div><br>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
Date: Tue, 31 Jul 2007 20:14:59 -0700
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
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	'Francois Audet' <audet@nortel.com>, Jon.Peterson@neustar.com,
	'Paul Kyzivat' <pkyzivat@cisco.com>
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The resource that was requested (sips:foo@example.com) is NOT  
available.  There may be a related but different resource available,  
so it is nice as a courtesy to provide this information, but the  
basic semantics should still be a 480.

thanks,
-rohan


On Jul 31, 2007, at 2:25 PM, Hadriel Kaplan wrote:

> Yeah but a 480 implies the resource is not available, whereas in  
> reality the
> resource *is* available using a different scheme.  I guess one  
> question is
> if the sips scheme literally means the UAS is a different  
> "resource".  If
> so, then registering a sips contact only should not implicitly make  
> the UAS
> available for routing plain sip.  The draft currently says they  
> represent
> the *same* resource.
>
> I'm not convinced by the downgrade argument.  The far-end can just  
> as easily
> send a 3xx redirecting the UAC to the sip contact.  Per this new  
> draft the
> proxies must not recurse on that, and the UAC should inform the  
> user before
> doing so.  But I think a new 4xx response code would be much more  
> likely to
> make that a reality, as any vendor implementing support for  
> recursing on a
> 418 would have read the draft and seen where it says "inform the  
> user".  If
> they didn't read the draft, they'd fail the request since they got an
> unknown 4xx response. (compare that to a 3xx, which is known/usable  
> without
> reading this draft)  Am I missing the gist of the concern?
>
> -hadriel
>
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@ekabal.com]
>> Sent: Tuesday, July 31, 2007 12:04 PM
>> To: Michael Thomas
>> Cc: Rohan Mahy; IETF SIP List; Francois Audet;  
>> Jon.Peterson@neustar.com;
>> Paul Kyzivat
>> Subject: Re: [Sip] draft-ietf-sip-sips-05: 480 vs. 418
>>
>> Michael,
>>
>> The semantics of a request to a sip: resource are different from a
>> request to a sips: resource.  If there are no Contacts registered
>> against a sips: resource, then the correct thing for the proxy to do
>> is to send a 480.  This is the same thing the proxy would do if there
>> were no Contacts registered against a sip: resource.
>>
>> We are not talking about a ciphersuite downgrade here. Providing an
>> error response code that encourages a client to downgrade is just
>> plain irresponsible.
>>
>> thanks,
>> -rohan
>>
>>
>> On Jul 27, 2007, at 9:39 AM, Michael Thomas wrote:
>>> Rohan Mahy wrote:
>>>> Michael,
>>>>
>>>> At issue here is what the default implementor is likely to do.
>>>> With a new 4xx, the misguided but well-meaning implementor is
>>>> likely to try to "helpfully" "repair" the error without thinking
>>>> about or understanding the security context.
>>>>
>>>> Using a Warning code raises the bar significantly, but still
>>>> allows automata to at least log what happened.
>>> As I said, a receiver is completely at liberty to prevent the
>>> downgrade by not
>>> accepting the downgraded request.  Problem solved by those who
>>> care. On
>>> the other hand I think it's pretty presumptuous that we "know" in
>>> all cases
>>> whether it's wrong to "repair" the error. Paul brought up several
>>> examples.
>>> Also: if the receiver allows non-SIPS requests why shouldn't you
>>> infer that
>>> they want you to "repair" the request? It sure seems like this is a
>>> misguided
>>> attempt at forcing ineffectual nannyware which is a pointless
>>> enterprise for
>>> IETF protocols.
>>>
>>>       Mike
>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip



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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



